| 00:00.07 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 00:00.32 | *** part/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 00:00.32 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 00:00.49 | surfdue | in gstreamer-properties when i change to alsa i hit test, and it says failed to construct pipeline |
| 00:01.28 | kyoo | Ariel_: What should I check next? |
| 00:01.40 | kyoo | What does DID mean? |
| 00:02.01 | DaPrivateer | direct inward dialing |
| 00:02.31 | *** join/#asterisk anthm[tablet] (i=anthm@208.254.19.131) |
| 00:02.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:02.48 | shido6 | need a DID? |
| 00:02.58 | shido6 | number for "ingress" or inbound calls |
| 00:03.05 | Ariel_ | kyoo, did you put the number at the end of your broadvoice account sip.broadvoice.com/number |
| 00:03.45 | kyoo | Ariel_: Yes, but I did not know what to pus, so I used my own extension, 305... |
| 00:03.51 | kyoo | s/pus/put/ |
| 00:04.01 | shido6 | err |
| 00:04.16 | shido6 | is 304 in "default" or the context you specified in [general] in sip.conf? |
| 00:04.20 | Ariel_ | kyoo, for broadvoice you need your phone number |
| 00:04.34 | surfdue | can asterisk transfer calles to other numbers |
| 00:04.37 | surfdue | like 1500 2020929 |
| 00:04.38 | surfdue | ? |
| 00:05.19 | Ariel_ | surfdue, like a number you dial out too? |
| 00:05.48 | kyoo | Ariel_: So it is: <mybvtelno>@sip.broadvoice.com:<mypass>:<mybvtelno>@sip.broadvoice.com/<mybvtelno> ? |
| 00:06.22 | Ariel_ | kyoo yes |
| 00:07.12 | surfdue | yes |
| 00:07.17 | surfdue | can u redirect it to another number? |
| 00:07.29 | Ariel_ | surfdue, yes with the right dial plan you can |
| 00:07.30 | kyoo | Ariel_: Still no change... |
| 00:07.38 | surfdue | dial plan? |
| 00:07.56 | Ariel_ | kyoo, are you using normal asterisk or something like amp added to your setup? |
| 00:08.09 | kyoo | Ariel_: I am using AMP via A@Home. |
| 00:08.45 | Ariel_ | kyoo, did you setup your did route? |
| 00:08.54 | surfdue | Ariel_ dial plan? |
| 00:09.01 | kyoo | Ariel_: No, I did not - I wasn't sure what it was for. |
| 00:09.26 | Ariel_ | kyoo, it's for routing your did's. (your phone number given you from bv) |
| 00:09.31 | nick125 | why doesnt asterisk start!?!? /me sighs |
| 00:09.49 | Ariel_ | nick125, something is missing or not configured correctly |
| 00:11.00 | kyoo | Ariel_: What should I put there? It wants DID number and destination... (reception (empty), extension, vm, ring group (empty), queue (empty), custom app, 'use incoming setting') |
| 00:11.18 | kyoo | Do I put my BV number in the number field? |
| 00:11.18 | nick125 | ah, i think some hold music i put in caused it o.0 |
| 00:11.33 | file[laptop] | Side effects may include bleeding from the eyes. |
| 00:11.42 | Ariel_ | kyoo, ok put your bv number then send it to were you want it to go too. like an extension.....hint |
| 00:11.49 | surfdue | nick |
| 00:11.57 | surfdue | HOLD MUSIC! |
| 00:11.57 | surfdue | Lol |
| 00:12.05 | kyoo | Ariel_: But this is for *outgoing* calls... Why would I want it to go to an extension? |
| 00:12.13 | nick125 | i removed the hold music, and it started |
| 00:12.39 | Ariel_ | kyoo, bv number is for incoming calls |
| 00:12.51 | Ariel_ | outgoing calls don't take this settings |
| 00:13.03 | Ariel_ | you need to create an outbound dialing rule. |
| 00:13.21 | kyoo | Ariel_: I'm trying to get outgoing calls working. I've set up an outbound routing... |
| 00:14.02 | Ariel_ | kyoo, you picked the broad voice trunk? |
| 00:14.12 | Ariel_ | when you dial out what does the CLI say |
| 00:14.49 | Ariel_ | nick125, are you using mpg123 or mpg321? |
| 00:15.14 | kyoo | Ariel_: Yes. the route is named "outgoingbv" no password, dial patterns 1NXXNXXXXXX NXXNXXXXXX NXXXXXX and trunk sequence 0 is SIP/BV1 ... |
| 00:16.42 | kyoo | Ariel_: I'm sorry - how do I know what the CLI says? |
| 00:16.56 | Ariel_ | well the last 2 should have a 1|NXXNXXXXXX and 1areacode|NXXXXXX |
| 00:17.35 | kyoo | Ariel_: AMP autogenerated those for me.. :) |
| 00:18.10 | Ariel_ | kyoo, yes but there dialing to a voip provider are they not? |
| 00:18.23 | kyoo | Ariel_: They are. |
| 00:19.42 | kyoo | The interface won't accept those numbers - it gives a DB error. |
| 00:22.13 | Nukemizer | any "bug track" admins here that can help me navigating for solution that Digium tech support directed me to look for ? |
| 00:22.18 | *** mode/#asterisk [+o twisted] by ChanServ |
| 00:23.26 | Nukemizer | I am looking for "ill allow you to assign "D" to the Hangup |
| 00:23.26 | Nukemizer | application" but am not having success locating in bug tracker |
| 00:23.48 | Nukemizer | "will allow you to assign "D" to the Hangup |
| 00:23.48 | Nukemizer | <PROTECTED> |
| 00:25.23 | Ariel_ | kyoo, you added something incorrectly then. or you tried to add bv dial route again. you need to clear that and edit the one you already have. |
| 00:26.37 | kyoo | Ariel_: This is what shows in the CLI when I attempt to call: http://rafb.net/paste/results/Yknpop52.html |
| 00:27.43 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 00:27.48 | kyoo | I use a full 10-digit number, which should get triggered by the 1NXXNXXXXXX, I think... But I do not see it getting triggered at all. |
| 00:27.58 | nick125 | hmm, when i try to login with kphone, it says forbidden, but, there isnt a place to put a password |
| 00:28.22 | jero | kyoo, did you put a _ at the beginning ? |
| 00:28.36 | kyoo | jero: At the beginning of ... ? |
| 00:28.46 | jero | of the expression |
| 00:29.01 | jero | like exten => _1NXXNXXXXXX,1,blah |
| 00:29.13 | Ariel_ | kyoo, change in your trunk from 0 for max channel to something like 2 or 3 |
| 00:29.29 | Ariel_ | jero, this is for out bound |
| 00:29.34 | jero | yes |
| 00:29.38 | kyoo | jero: I'm using AMP - I tmay do that part for me, but it's examples do not use that. |
| 00:29.38 | Ariel_ | jero, and it's via amp |
| 00:29.45 | jero | oh ok sorry |
| 00:30.05 | Ariel_ | kyoo, your report says you dont have enough channels available and it's set to 0 |
| 00:30.13 | jero | amp has 2 places to define that, you may be in the wrong one |
| 00:30.40 | kyoo | Ariel_: That looks like it ... IT IS! |
| 00:30.55 | Ariel_ | Jero correct it does have 2 places |
| 00:31.27 | jero | and the two places interact quite a bit, I dont remember how, didnt use it for something like 8 months |
| 00:31.46 | kyoo | Ariel_: The interface says "blank for unlimited" and it defaults to 0. oops. :) |
| 00:31.52 | kyoo | Ariel_: Thanks so much. |
| 00:32.19 | Ariel_ | kyoo, no problem |
| 00:32.22 | jero | I'll try to package sflphone for many linux distros and other oses eventually |
| 00:32.31 | Ariel_ | jero it's changed in 8 months allot |
| 00:32.49 | jero | Ariel_, theres no doubt :) |
| 00:32.54 | jero | anyone ever tried sflphone ? |
| 00:33.37 | nick125 | ok i just created a extendion on AMP, how do i get x-lite to use it? |
| 00:33.55 | jero | log in your xlite as this extension |
| 00:34.03 | tzanger | I love the 'babysitting email scanner' someone has on -users |
| 00:34.18 | tzanger | my posts to the list were automaticlaly deleted because of sexual discrimination... ?? |
| 00:34.27 | tzanger | I love the us legal system that makes this kind of thing necessary |
| 00:34.39 | nick125 | jero: i try to, using the correct pass and everything, and it says forbidden |
| 00:35.09 | jero | hrm |
| 00:35.42 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
| 00:35.45 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-71-65-37-102.indy.res.rr.com) |
| 00:37.02 | Ariel_ | nick125, if it's saying forbidden then there is something incorreclty set up on xlite |
| 00:38.01 | Ariel_ | tzanger, live is so strange. Sex is all over the place it's just part of life....some do some don't |
| 00:38.09 | Ariel_ | live/life |
| 00:39.20 | tzanger | yeah |
| 00:39.20 | jero | nite all |
| 00:40.05 | *** join/#asterisk otaku_p0pe (i=foobar@blk-222-96-154.eastlink.ca) |
| 00:40.11 | otaku_p0pe | oh wow |
| 00:40.27 | Ariel_ | jero, goodnight |
| 00:40.41 | otaku_p0pe | so i bet this question gets asked all the time. |
| 00:40.45 | otaku_p0pe | anyone use asterisk + broadvoice ? |
| 00:41.04 | *** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 00:41.17 | Ariel_ | otaku_p0pe, yes |
| 00:41.44 | otaku_p0pe | i keep getting 401 unauthorized from their sip gateway. now i know the password and such is correct |
| 00:41.50 | Ariel_ | oh it's yes it's asked allot and yes people use it. |
| 00:42.00 | otaku_p0pe | the only things i can think of is the NAT is causing trouble, or it takes a few hours to actually activate the account |
| 00:42.08 | otaku_p0pe | note : i've gotten the email saying my account is active |
| 00:42.12 | otaku_p0pe | so i dunno. |
| 00:42.31 | *** join/#asterisk asteriskjohn (n=johnb@ip24-251-151-16.ph.ph.cox.net) |
| 00:42.34 | Ariel_ | otaku_p0pe, are you putting the account to an ata and not to an asterisk box? |
| 00:42.53 | otaku_p0pe | to an asterisk box. |
| 00:43.09 | otaku_p0pe | the asterisk box works - i have an FXS / softphones that work gloriously. except over gprs. |
| 00:43.25 | bkw__ | it won't |
| 00:43.35 | bkw__ | grps is too slow... latent and sucks |
| 00:43.38 | otaku_p0pe | i know. |
| 00:43.48 | otaku_p0pe | well, it's just latency |
| 00:43.52 | kyoo | Ariel_: Can you give any suggestions about incoming on the same trunk? (it goes directly to BV voicemail, and never even shows in the CLI sip debug...) |
| 00:44.02 | otaku_p0pe | i got 35k/s on my gprs one time doing apt-get upgrade (don't ask) |
| 00:44.42 | Ariel_ | kyoo, remember the did routes |
| 00:45.05 | kyoo | Ariel_: I have set up a DID route, to the receptionist... |
| 00:45.32 | otaku_p0pe | bleh |
| 00:45.34 | kyoo | The DID number is simple my BV number, without the :1:, correct? |
| 00:45.38 | Ariel_ | kyoo, you have it registered correct? |
| 00:45.46 | Ariel_ | yes |
| 00:45.56 | kyoo | Ariel_: As far as I know - it reports registered. |
| 00:46.16 | nick125 | Ariel_: i cant find anythong incorrectly setup in xlite.. |
| 00:46.23 | otaku_p0pe | i wish mine reported registered :( |
| 00:46.42 | *** join/#asterisk brimston1 (n=brimston@pcp01534724pcs.huntsv01.al.comcast.net) |
| 00:46.50 | Ariel_ | nick125, are you on the same network? |
| 00:46.57 | nick125 | yeah |
| 00:47.32 | nick125 | my ip is 1.9, server is 1.15 |
| 00:47.41 | nick125 | (all in private block) |
| 00:47.44 | Ariel_ | nick125, what does the cli say wen your trying to register |
| 00:47.57 | kyoo | Ariel_: It actually shows 2 SIP peers for bv - bv1/<my bv number> and <my bv number>/6034, both connected to the same BV host. it that normal? |
| 00:48.13 | nick125 | the asterisk cli? |
| 00:48.36 | Ariel_ | kyoo, no |
| 00:48.55 | Ariel_ | nick125, yes |
| 00:49.04 | nick125 | if i try to make a call out to something like *411, it says 'service unavailable' in the xlite logs |
| 00:49.09 | Ariel_ | kyoo, you might have added two trunks |
| 00:49.15 | nick125 | nope, dont see anything |
| 00:49.33 | otaku_p0pe | man oh man |
| 00:49.38 | Ariel_ | f9 on xlite gives you a debug settings |
| 00:49.57 | kyoo | Ariel_: I don't think so ( no second one is showing) but it may be because it have the entire set of BV connection information in each of the textareas on the trunk setup - Outgoing and Incoming... |
| 00:50.01 | nick125 | thats what ive been looking at to find the service unavailable |
| 00:50.39 | nick125 | hmm, in amp, what type should i set? |
| 00:50.51 | nick125 | oh nvm |
| 00:51.09 | otaku_p0pe | Ariel_: do you use AMP ? |
| 00:51.23 | nick125 | wait, in the diagnostics, aint that the raw stuff of something? |
| 00:51.28 | nick125 | i dont see it sending the password |
| 00:52.03 | Ariel_ | otaku_p0pe, yes and no. I use it for some of my customers |
| 00:52.11 | otaku_p0pe | with broadvoice ? |
| 00:52.12 | Ariel_ | I don't use it for my self |
| 00:52.16 | Ariel_ | yes |
| 00:52.33 | otaku_p0pe | hmm. i'm betting it's my register line that's messed |
| 00:52.40 | Ariel_ | nick125, hummm could it be you did not set it up correctly. |
| 00:52.42 | kyoo | Ariel_: What information goes in the "Incoming" textarea? |
| 00:52.53 | otaku_p0pe | it's number@sip.broadvoice:password:number@sip.broadvoice/number ? |
| 00:53.05 | Ariel_ | otaku_p0pe, yes |
| 00:53.36 | nick125 | Ariel_: i set it up as this: username/authorization: 200 |
| 00:53.40 | otaku_p0pe | ugh. i really really really do not want to strace asterisk |
| 00:53.43 | Ariel_ | incoming text area???? |
| 00:53.44 | nick125 | Password: <my password> |
| 00:54.08 | nick125 | Domain/Realm and SIP Proxy: <my asterisk box> |
| 00:54.20 | kyoo | Ariel_: On the trunk setup, there is a textarea called "incoming" wher eyou put all of the SIP settings for incoming calls. Should that be a copy of the outgoing box? |
| 00:54.48 | otaku_p0pe | kyoo: are you also using AMP / broadvoice ? |
| 00:54.55 | otaku_p0pe | and no, it's not an exact copy. |
| 00:54.56 | kyoo | otaku_p0pe: yes. |
| 00:54.56 | *** join/#asterisk zox (n=zox@ip70-176-64-134.ph.ph.cox.net) |
| 00:55.04 | Ariel_ | kyoo, oh I seee you did not set the trunk up correctly. |
| 00:55.31 | otaku_p0pe | kyoo: http://geekgazette.com/index.php?option=com_content&task=view&id=20&Itemid=26 |
| 00:55.34 | Ariel_ | ok everyone with amp and needs help lets go to the #amportal where there are others there that use it as well. |
| 00:55.34 | otaku_p0pe | that's what i used |
| 00:55.35 | kyoo | Ariel_: It would not surprise me. I don't know what to put in the trunk boxes, so I copied the setup info from bv.com's asterisk page... |
| 00:55.51 | otaku_p0pe | but my trunk isn't registering correctly with BV's sip proxy :/ |
| 00:55.55 | Ariel_ | kyoo, no don't one is peer the other is user |
| 00:55.58 | Ariel_ | for type |
| 00:56.17 | Ariel_ | argh I will be back in a few minutes.... I need coffeee........ |
| 00:56.20 | otaku_p0pe | hehe. |
| 00:56.35 | nick125 | btw, how do i tell if amp correctly added the sip user? |
| 00:56.36 | Ariel_ | lets all take this to the #amportal |
| 00:57.09 | Ariel_ | nick125, you can view the sip_additional.conf file |
| 00:57.48 | nick125 | there isnt a sip_additional.con file :/ |
| 00:58.11 | nick125 | conf |
| 00:59.28 | nick125 | let me guess, there should be one of these files, correct? |
| 00:59.48 | *** join/#asterisk mago3-cn (i=maxgluck@200.109.166.172) |
| 01:00.16 | Ariel_ | nick125, please go do a little reading here is a page that has amp and xlite on it with pictures....http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm |
| 01:00.20 | kyoo | otaku_p0pe: Ariel_: Using the page at geekgazette, it is not working. Thank you both very much. |
| 01:00.26 | kyoo | s/not/now/ |
| 01:01.00 | surfdue | trust me nick is trying his hardest |
| 01:01.01 | surfdue | on his own |
| 01:01.03 | surfdue | i can hear him |
| 01:01.42 | otaku_p0pe | kyoo: what isn't working ? |
| 01:01.44 | otaku_p0pe | wait. it is working ? |
| 01:01.45 | otaku_p0pe | dammit |
| 01:01.49 | otaku_p0pe | heh. it's not for me :/ |
| 01:01.53 | Ariel_ | nick125, if the file is not there and not being writen when you create the sip extension and pressed the red bar then you have rights problems |
| 01:02.04 | kyoo | otaku_p0pe: it is working! |
| 01:02.08 | otaku_p0pe | good ! |
| 01:02.21 | kyoo | otaku_p0pe: What part is not nowring for you? In or Out? |
| 01:02.25 | otaku_p0pe | either |
| 01:02.28 | otaku_p0pe | it can't register with BV |
| 01:02.53 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 01:02.55 | kyoo | otaku_p0pe: You have had BV provision you a generic device? |
| 01:03.53 | otaku_p0pe | that's what they said. |
| 01:04.17 | otaku_p0pe | oh hell |
| 01:04.28 | file[laptop] | brimstone: Matt! |
| 01:04.37 | nick125 | ah |
| 01:04.40 | nick125 | found the problem |
| 01:04.43 | nick125 | permissions |
| 01:04.52 | brimstone | werd up file[laptop] |
| 01:04.59 | kyoo | Essentially, registering was the easiest part for me: Provision with BV, get then to mail me the password, and set up the trunk so it regestered with mynumber@sip.broadvoice.com:secret:mynumber@sip.broadvoice.com, and then set up sip.broadvoice.com in my hosts file... |
| 01:05.17 | otaku_p0pe | haha hell. |
| 01:05.23 | *** join/#asterisk reagent (i=mathias@2002:d4fe:b1a1:0:20d:88ff:fef4:66bb) |
| 01:05.24 | otaku_p0pe | so i use this password they generate for me; not my login |
| 01:05.44 | kyoo | otaku_p0pe: Correct! |
| 01:06.37 | reagent | hi. I set up askterisk as a SIP client (connecting to a SIP server and waiting for calls from there). When I run asterisk, it tells me that this channel is unreachable. How can I find out why? |
| 01:06.49 | reagent | it says: Peer '41325110504' is now UNREACHABLE! |
| 01:07.03 | nick125 | whats the number to test asterisk IAX? |
| 01:07.10 | bkw__ | reagent, any number of things can cause that |
| 01:07.17 | bkw__ | but don't take that as the TRUTH |
| 01:07.33 | reagent | bkw__: Hm |
| 01:07.52 | bkw__ | thats just a sip packet that didn't make it back or the end point didn't respond.. or went braindead |
| 01:07.54 | bkw__ | its not a ping |
| 01:08.00 | bkw__ | people seem to think that qualify is a ping |
| 01:08.26 | Ariel_ | nick125, you mean 500 in the sample setup files |
| 01:08.35 | reagent | bkw__: Hm, maybe because it couldn't properly register? |
| 01:08.36 | brimstone | file[laptop]: what are you upto? |
| 01:08.39 | nick125 | ah yes |
| 01:08.42 | file[laptop] | brimstone: download tunes! |
| 01:08.49 | bkw__ | reagent, could be |
| 01:08.50 | brimstone | download? |
| 01:09.04 | nick125 | Ariel_: what file do i enable it in again? |
| 01:09.10 | otaku_p0pe | hmm. so close i can taste it. |
| 01:09.18 | file[laptop] | downloading |
| 01:09.24 | brimstone | ooooohhhhh |
| 01:09.32 | reagent | bkw__: I just saw with SIP SHOW REGISTRY that in waits a long time while 'Packet Sent' and then the registration timeouts |
| 01:11.17 | reagent | bkw__: I can connect to the same proxy with xlite... something must be wrong :) |
| 01:11.33 | adelas | hey how can i tell if my cisco 7960 phone is connecting to my pbx server? |
| 01:11.39 | kyoo | IAX is more robust and efficient than SIP? |
| 01:11.54 | bkw__ | depends |
| 01:12.00 | bkw__ | that question would start a ware |
| 01:12.05 | bkw__ | SIP scales better than IAX |
| 01:12.11 | bkw__ | but that all depends on usage |
| 01:12.17 | bkw__ | in some cases iax will win |
| 01:12.34 | kyoo | For small call volume. (2 calls simultaneous max) |
| 01:12.56 | bkw__ | use sip |
| 01:13.02 | bkw__ | unless you have nat |
| 01:13.04 | bkw__ | then use iax2 |
| 01:13.11 | bkw__ | but in small volume its really up to you |
| 01:13.23 | DarthClue | iax/iax2 has issues that sip doesn't, sip has issues that iax/iax2 doesn't. use what works best. |
| 01:14.02 | Ariel_ | wow DarthClue well said |
| 01:14.03 | kyoo | bkw_: NAT at both ends. Essentially, wondering if SIP to VOIP provider to PSTN or IAX2 direct to IAX2 is going to be better use of limited bandwidth... |
| 01:14.07 | Ariel_ | ok got my coffee. |
| 01:14.23 | Ariel_ | kyoo, nat both ends iax2 is better |
| 01:14.30 | DarthClue | Ariel_: i'm barely awake, surprised i can even comprehend anything at this point. |
| 01:14.32 | Ariel_ | but if done correctly sip will work. |
| 01:14.50 | bkw__ | kyoo, use IAx2 |
| 01:14.57 | kyoo | Ariel_: Which one will cope better with limited bandwidth? (NO QOS routers in place yet.) |
| 01:15.10 | Qwell | and with iax2, you can trunk |
| 01:15.10 | Ariel_ | kyoo, none |
| 01:15.16 | Qwell | nice little benefit |
| 01:15.20 | Ariel_ | but iax2 will work better via dual nat |
| 01:15.25 | kyoo | Qwell: What does that mean? |
| 01:15.31 | DarthClue | if you have the exact same version of iax2 on each box. |
| 01:15.40 | Qwell | DarthClue: oh? |
| 01:15.50 | kyoo | DarthClue: They will be the same version of A@H. :) |
| 01:15.56 | Ariel_ | lets not mix things up please |
| 01:16.29 | Ariel_ | kyoo, if your talking between two asterisk boxes use iax2 |
| 01:16.51 | otaku_p0pe | this is kinda cute. |
| 01:17.24 | *** join/#asterisk L|NUX (n=linux@202.5.145.14) |
| 01:17.27 | otaku_p0pe | it just rings forever. |
| 01:17.31 | otaku_p0pe | on outgoing. |
| 01:17.35 | kyoo | Ariel_: I'm debating ig it is worth installing a second asterisk box. (They already have a PBX on location, but if it will help with bandwidth issues, I will attach a * box to the pbx for the time being...) |
| 01:17.52 | kyoo | otaku_p0pe: You have registration now? |
| 01:17.56 | DarthClue | Qwell: if you don't have the same version of iax2 on each box you will see issues. sometimes consistent, sometimes random. |
| 01:17.56 | otaku_p0pe | kyoo: yup |
| 01:18.01 | otaku_p0pe | i was using the wrong pw ^_^ |
| 01:18.09 | kyoo | otaku_p0pe: Was it the "special" password? |
| 01:18.13 | kyoo | otaku_p0pe: yup! |
| 01:18.13 | otaku_p0pe | yeah :( |
| 01:18.37 | kyoo | otaku_p0pe: From there, your "geek" instructions worked perfectly. |
| 01:18.47 | Ariel_ | kyoo, well the bandwidth issue depends more on the codec your using. not if it's iax2 or sip. |
| 01:18.48 | otaku_p0pe | heh. |
| 01:18.57 | nick125 | in my firewall, what ports should i open up so other people can login to the system? |
| 01:19.00 | Ariel_ | But if your trunking more then 3 calls on one connection iax2 is better. |
| 01:19.19 | Ariel_ | 5060/61 and 10,000 to 20,000 for sip |
| 01:19.25 | kyoo | Ariel_: Oh, of course. Can I choose the codec, or does my VOIP provider or FXS do that for me? |
| 01:20.02 | Ariel_ | kyoo, depends on provider bv only supports ulaw. But between your own box you can do what codec you have. |
| 01:20.08 | otaku_p0pe | oh. do i need a DID route for the outgoing ? or incoming ? or am i drunk again ? |
| 01:20.36 | kyoo | Ariel_: Do other providers provide other codecs, or is ulaw a standard? |
| 01:21.08 | Ariel_ | kyoo, others do provide other codec. But ulaw is the best due to sound. But it takes the most bandwidth. |
| 01:21.57 | kyoo | Ariel_: I heard somthing about a g-something-7-something codec in here earlier. It is claimed to sound as good, but much better on bandwidth? |
| 01:22.13 | Qwell | g729 |
| 01:22.14 | nick125 | 5060/61? |
| 01:22.14 | adelas | hey dose anyone know a way to clear out like the asterisk in system, so that i can reinstall it and start over? |
| 01:22.16 | nick125 | oh |
| 01:22.16 | nick125 | nvm |
| 01:22.22 | otaku_p0pe | oh this is odd |
| 01:22.32 | otaku_p0pe | so it will ring, but when the other party answers the connection craps out |
| 01:22.38 | kyoo | Qwell: Are there voip providers I can use g729 with? |
| 01:22.43 | Qwell | sure |
| 01:22.55 | otaku_p0pe | isn't g729 a pain with asterisk ? |
| 01:22.55 | kyoo | Qwell: Are thery considerably more expensive? ;) |
| 01:22.59 | Ariel_ | rm -rf /var/lib/asterisk/modules |
| 01:22.59 | Qwell | dunno |
| 01:23.12 | otaku_p0pe | Ariel_: hee. |
| 01:23.17 | nick125 | Ariel_: udp or tcp? |
| 01:23.21 | Ariel_ | g729 is great codec. it's extra cost with asterisk it's not freeeee |
| 01:23.26 | kyoo | Ariel_: That sounds like a dangerous command to just throw out here. :) |
| 01:23.29 | Ariel_ | udp |
| 01:23.33 | nick125 | ah ok |
| 01:23.44 | Ariel_ | they asked |
| 01:23.51 | kyoo | Ariel_: Oh, so they did. :) |
| 01:24.11 | *** join/#asterisk grimse_ (n=grimse@p5481EF73.dip.t-dialin.net) |
| 01:24.15 | Ariel_ | now it's up to people to use it correctly just a gun. The gun does not kill it's the person firing it. |
| 01:25.01 | otaku_p0pe | hahaha. |
| 01:25.35 | Druken | it's actually the bullet that kills |
| 01:25.50 | Ariel_ | Druken, Symantec |
| 01:25.54 | Druken | :) |
| 01:25.58 | otaku_p0pe | wait. broadvoice claims i don't need to open anything in my firewall |
| 01:26.01 | otaku_p0pe | are they lying to me ? |
| 01:26.07 | Qwell | otaku_p0pe: usually |
| 01:26.07 | Druken | unless you beat them with the gun |
| 01:26.09 | Ariel_ | lier lier |
| 01:26.14 | kyoo | otaku_p0pe: I have nothing open. |
| 01:26.16 | otaku_p0pe | cause that would explain the issue with it crapping out when they other party answers |
| 01:26.19 | otaku_p0pe | kyoo: does outgoign work fine ? |
| 01:26.19 | leoncamel | good morning . :) |
| 01:26.22 | kyoo | otaku_p0pe: (related to this anyway. :) ) |
| 01:26.40 | kyoo | otaku_p0pe: yes, perfectly. (A little quiet...) |
| 01:26.43 | otaku_p0pe | heh. |
| 01:26.45 | Qwell | otaku_p0pe: If your router doesn't pass related packets, it'll break entirely |
| 01:26.54 | otaku_p0pe | Qwell: like it won't ring even. |
| 01:26.55 | otaku_p0pe | ? |
| 01:27.04 | Qwell | nothing will happen |
| 01:27.08 | Ariel_ | if something does not see it how can it ring |
| 01:27.14 | otaku_p0pe | yeah exactly. |
| 01:27.17 | otaku_p0pe | ok. so it's something else. |
| 01:27.53 | otaku_p0pe | wierd. |
| 01:28.58 | otaku_p0pe | i know. i bet i need to have some sort of human sacrifice going on. |
| 01:29.15 | Qwell | What? You didn't sacrifice a sheep? |
| 01:29.24 | Qwell | its in the install instructions... |
| 01:29.26 | otaku_p0pe | no. just a lowly virgin. |
| 01:29.29 | otaku_p0pe | :( |
| 01:29.42 | Qwell | those aren't mutually exclusive |
| 01:29.47 | surfdue | hey i cant talk in xlite |
| 01:29.49 | surfdue | anyone know why |
| 01:29.50 | Ariel_ | virgin's you mean there are still some available |
| 01:29.51 | otaku_p0pe | hmmmm. |
| 01:30.01 | otaku_p0pe | surfdue: yes. something is not configured correctly. |
| 01:30.06 | Ariel_ | xlite not working correctly. |
| 01:30.11 | surfdue | can anyone help? |
| 01:30.13 | otaku_p0pe | xlite was a pain for me to get working. |
| 01:30.20 | Ariel_ | xlite is easy |
| 01:30.30 | otaku_p0pe | Ariel_: yes but you appear to be a ninja. |
| 01:30.35 | surfdue | um |
| 01:30.37 | surfdue | i can hear |
| 01:30.39 | surfdue | and everything |
| 01:30.44 | surfdue | but he cant hear me |
| 01:30.50 | surfdue | and i can see my bar goes up as i talk 2 |
| 01:31.09 | Druken | virgins? ya 10 and under,... and sometimes not even then... |
| 01:31.14 | otaku_p0pe | Ariel_: hee |
| 01:31.34 | twisted | wtf |
| 01:31.38 | *** join/#asterisk Barmal (n=info@c-24-92-153-118.hsd1.ga.comcast.net) |
| 01:31.48 | otaku_p0pe | Ariel_: any ideas on the issue i'm having though ? or places i could check for problems ? |
| 01:31.52 | Ariel_ | nat |
| 01:31.58 | otaku_p0pe | yeah it is nat'd |
| 01:31.59 | otaku_p0pe | hmm. |
| 01:32.08 | otaku_p0pe | i added an exter_ip thing in sip.conf |
| 01:32.20 | otaku_p0pe | maybe port-forward back in ? |
| 01:32.24 | Ariel_ | externip= |
| 01:32.25 | Druken | nat can be such a pain |
| 01:32.26 | otaku_p0pe | ahh tcpdump. why didn't i think of you sooner ? |
| 01:33.25 | Ariel_ | otaku_p0pe, it uses udp |
| 01:34.04 | Ariel_ | otaku_p0pe, most of the sound ports are 10,000 to 20,000 rtp for sip |
| 01:34.38 | nick125 | wee :) |
| 01:34.41 | otaku_p0pe | oi vei |
| 01:34.44 | nick125 | my asterisk works :) |
| 01:34.50 | otaku_p0pe | so i have to like forward all of those ? that sounds odd. |
| 01:34.52 | otaku_p0pe | but hey |
| 01:35.29 | Ariel_ | well sip uses them for sound iax2 only uses 4569 for sound and registration. |
| 01:36.01 | *** join/#asterisk tengulre (n=tengulre@61.185.238.166) |
| 01:36.08 | otaku_p0pe | yeah. |
| 01:36.11 | otaku_p0pe | <3 iax2 |
| 01:36.43 | otaku_p0pe | yeah that's it |
| 01:36.49 | otaku_p0pe | i have to forward 10k-20k to my pbx |
| 01:36.49 | otaku_p0pe | ugh |
| 01:37.04 | otaku_p0pe | that is =ridiculous= |
| 01:37.05 | Ariel_ | you want to hear them don't you |
| 01:37.09 | otaku_p0pe | yeah. |
| 01:37.12 | otaku_p0pe | but guh. |
| 01:37.37 | otaku_p0pe | i think that once i move i'll setup the pbx on a public ip, only with a bridged firewall between it and the internets |
| 01:37.39 | Ariel_ | if you don't have too many sip users you can cut that down and edit the rtp.conf file |
| 01:37.46 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 01:37.46 | *** mode/#asterisk [+o bkw__] by ChanServ |
| 01:37.58 | otaku_p0pe | yeah. |
| 01:38.36 | Ariel_ | ok so now we have kyoo, nick125 and otaku_p0pe working finally????? |
| 01:38.44 | otaku_p0pe | yeah ! |
| 01:39.22 | otaku_p0pe | sweeet. |
| 01:39.23 | Ariel_ | NEXT! NEXT! I take donations at arielb27@gmail.com paypal hehehehehe |
| 01:39.32 | *** join/#asterisk santiago (n=santiago@63.245.86.202) |
| 01:39.41 | otaku_p0pe | hey if i wasn't so poor my only phone line was a BYOD Lite i would totally do that Ariel_ |
| 01:39.44 | otaku_p0pe | =totally= |
| 01:40.27 | otaku_p0pe | oh i like how you =cau= use multiple SIP connections with broadvoice. just they charge you for it |
| 01:40.56 | Druken | network engineer, that's as bad as a starving artist... |
| 01:40.59 | Ariel_ | they allow you to have 2 channels with them. |
| 01:41.23 | Ariel_ | Druken, yep but I have an house full of girls to feed. |
| 01:41.44 | Druken | so you really are starving... :) |
| 01:42.14 | otaku_p0pe | ok i have some perfomance tuning |
| 01:42.36 | *** join/#asterisk pbd (n=pbdavids@12.144.118.37) |
| 01:42.40 | otaku_p0pe | but it's mostly due to the fact that osX is a memory hog and i only have 512m of ram :/ |
| 01:42.52 | pbd | Even', all. |
| 01:43.06 | Ariel_ | otaku_p0pe, you can do noload on what you don't need in asterisk. |
| 01:43.18 | Ariel_ | look at modules.conf |
| 01:43.28 | Ariel_ | pbd, hello |
| 01:43.50 | otaku_p0pe | yeah that might help too |
| 01:43.51 | pbd | Anyone out there seen a case of one-way DTMF tones on an h.323 (nufone) channel? (Head, of earlier this week) |
| 01:44.27 | pbd | RTP is fine, signalling is fine- asterisk RECEIVEs the oob dtmf, but doesn't SEND it. |
| 01:44.32 | surfdue | whats the extention to dial to get the main menu u setup |
| 01:44.51 | Ariel_ | surfdue, amp? |
| 01:44.52 | nick125 | if i want to get to the main menu (digital receptionist), how do i access that from the internal network (off my sip phone)? |
| 01:44.55 | Druken | does anyone know the guidelines for any other numbering plans than NANPA ? |
| 01:44.57 | pbd | Ariel- good choice. Unfortunately, I'm stuck with it- integrating with Cisco Callmanager. |
| 01:45.15 | Ariel_ | argh dtmfmode=inband |
| 01:45.15 | Druken | H323 is EVIL! |
| 01:45.23 | Ariel_ | h323 is not evil |
| 01:45.32 | pbd | (and yes, I know I can use SIP- I do, and it works fine- but CCM only does compressed codecs with h.323, not SIP) |
| 01:45.35 | Druken | is in my mind... |
| 01:45.42 | Ariel_ | digital recp 777 in amp |
| 01:45.45 | pbd | Can't use inband- I'll be using 729 in the final mix. |
| 01:45.46 | nick125 | is there an extension to access it? |
| 01:45.59 | Ariel_ | rfc2833 |
| 01:46.20 | pbd | I've got that set- and it works, INBOUND... but doesn't send it. Codecs appear negotiated correctly. |
| 01:46.31 | pbd | I'm halfway to filing a bug on mantis. |
| 01:46.37 | Druken | nick125: try dialing the main incoming number? |
| 01:46.41 | otaku_p0pe | pdb : i read that as martinis |
| 01:46.45 | Ariel_ | pbd, there is a channel called #nufone |
| 01:47.11 | Ariel_ | no one seen what I typed 777 |
| 01:47.17 | nick125_lappy | Druken, what if its a pstn number? |
| 01:47.24 | otaku_p0pe | so incoming doesn't work. |
| 01:47.30 | otaku_p0pe | <3 rfc's |
| 01:47.31 | nick125_lappy | but, the pstn number isnt setup yet |
| 01:47.50 | pbd | Yeah, but that's for nufone's business services.. this is an asterisk channel driver issue, and, like it or not, nufone's channel driver is what's packaged in CVS- and the one I've found actually works best. |
| 01:47.54 | Druken | nick125: well if your dialplan is done correctly, it'll route it to where it needs to go |
| 01:48.19 | nick125_lappy | ooo, would DID routes work? |
| 01:48.31 | nick125_lappy | get the DID number to something like *10, and forward it to the menu? |
| 01:48.48 | nick125_lappy | will that work from the internal phones? |
| 01:48.53 | *** part/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 01:49.15 | Ariel_ | nick125_lappy, in what program? |
| 01:49.38 | nick125 | DID in AMP, using xlite as a sip client to my asterisk |
| 01:50.38 | Ariel_ | nick125, if you dial exten 777 it takes you to the digital recp. |
| 01:51.13 | *** part/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 01:51.22 | Ariel_ | nick125, did route work if you have a trunk for inbound setup |
| 01:51.28 | *** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 01:51.31 | nick125 | so, i would type '777' into xlite, and it would treat it as a incoming pstn call? |
| 01:51.36 | *** part/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 01:51.44 | *** join/#asterisk TeknoKosh (n=evangeli@ip157.net65.ipnetworks.net.au) |
| 01:51.47 | Ariel_ | nick125, yes |
| 01:51.52 | *** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net) |
| 01:51.56 | nick125_lappy | neato |
| 01:52.36 | nick125 | does asterisk have like a guide for all these codes and everything lol |
| 01:52.43 | otaku_p0pe | what does BV need udp:69 for ? |
| 01:52.47 | Ariel_ | rofl |
| 01:52.51 | Ariel_ | ~doc |
| 01:52.51 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 01:52.51 | Qwell | tftp? |
| 01:53.10 | *** join/#asterisk DarthClue (n=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
| 01:54.49 | Ariel_ | otaku_p0pe, that is for if you use an ata with them. |
| 01:55.01 | otaku_p0pe | oh ok. |
| 01:55.05 | otaku_p0pe | hmmm. |
| 01:55.14 | Ariel_ | you don't need it with an asterisk box |
| 01:55.49 | otaku_p0pe | so who wants to call my number |
| 01:56.27 | *** join/#asterisk Chuji (i=Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
| 01:57.51 | *** join/#asterisk BrainSurg (n=paul@69.158.227.13) |
| 01:58.03 | BrainSurg | Greetings. |
| 01:58.08 | otaku_p0pe | hi |
| 01:58.41 | BrainSurg | otaku_p0pe: How goes it? |
| 01:58.47 | otaku_p0pe | not too bad. |
| 01:58.56 | otaku_p0pe | just fighting with incoming on my shiny new sip trunk |
| 01:59.25 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 02:00.17 | BrainSurg | nifty. |
| 02:00.25 | *** join/#asterisk Tili (i=Tili@219.137.38.125) |
| 02:00.31 | BrainSurg | Got any pointers to setting up a basic asterisk config? |
| 02:00.39 | BrainSurg | I'm just starting to play with it. |
| 02:01.42 | otaku_p0pe | yes. use the asterisk management portal |
| 02:01.46 | otaku_p0pe | or just go for asterisk@home |
| 02:01.52 | otaku_p0pe | that's what iw oudl have done, but i didn't have a spare pc |
| 02:01.55 | pbd | Brain: Spend a few quality hours with www.voip-info.org |
| 02:02.02 | otaku_p0pe | which is somewhat annoying |
| 02:02.25 | krisguy | I'm glad govt auctions happen quite a bit around here |
| 02:02.39 | krisguy | great way to get spare boxen cheap |
| 02:02.44 | otaku_p0pe | yeah |
| 02:02.47 | otaku_p0pe | i actually have a spare box now |
| 02:02.49 | Ariel_ | ebay |
| 02:02.52 | otaku_p0pe | but no time to swap the system :D |
| 02:02.57 | otaku_p0pe | Ariel_: see previous statement about being broke |
| 02:03.10 | krisguy | I started with Asterisk@Home, good way to get wet |
| 02:03.11 | Ariel_ | otaku_p0pe, so am i |
| 02:03.16 | otaku_p0pe | hehe |
| 02:03.21 | Ariel_ | so what did you use vmware? |
| 02:03.26 | BrainSurg | krisguy: What is Asterisk@Home? |
| 02:03.36 | *** join/#asterisk wulfy814 (n=lorentz@c-67-165-37-20.hsd1.pa.comcast.net) |
| 02:04.03 | Ariel_ | BrainSurg, it's a ISO that comes with OS , asterisk and a nice gui and many more nice things for asterisk. |
| 02:04.08 | krisguy | ISO with CentOS, Asterisk, MySQL, etc. |
| 02:04.13 | BrainSurg | hmm. |
| 02:04.24 | nick125 | since sometimes a softphone is kinda diffcult |
| 02:04.27 | BrainSurg | Got Ubuntu running with asterisk installed already:) |
| 02:04.35 | krisguy | nick125, shouldn't be a problem |
| 02:04.36 | Ariel_ | nick125, yes |
| 02:04.47 | Tili | is the challenge sent by asterisk always the same or something random |
| 02:04.53 | Ariel_ | BrainSurg, ok then see the docs and make samples |
| 02:04.53 | krisguy | ATA boxen will work with Asterisk |
| 02:04.58 | Ariel_ | ~doc |
| 02:04.58 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 02:05.25 | BrainSurg | jbot: thanks. |
| 02:05.25 | jbot | my pleasure, BrainSurg |
| 02:05.50 | krisguy | BrainSurg: voip-info.org is a godsend |
| 02:05.57 | BrainSurg | I'll check it out. |
| 02:06.07 | wulfy814 | BrainSurg: how about "make config" it didn't work for me in Ubuntu |
| 02:06.36 | nick125 | i might see if i can find someone with a ATA box or buy one on ebay or something |
| 02:06.38 | Ariel_ | wulfy814, that is due to make config is more for rh type of system. ubuntu is debian based |
| 02:07.00 | *** join/#asterisk santiago (n=santiago@63.245.86.202) |
| 02:07.00 | Ariel_ | nick125, make sure it's unlocked one |
| 02:07.23 | BrainSurg | wulfy814: I just put asterisk on using apt. |
| 02:07.35 | Ariel_ | rumor has it the adp-get has one some place |
| 02:07.51 | Ariel_ | apt-get |
| 02:07.58 | otaku_p0pe | hmm |
| 02:08.07 | Ariel_ | argh I need to go to bed.... coffee did not help. |
| 02:08.15 | otaku_p0pe | would a pIII/500 with 256m be good enough for asterisk ? |
| 02:08.20 | Ariel_ | yes |
| 02:08.24 | otaku_p0pe | thought so |
| 02:08.35 | otaku_p0pe | yeah. so i just have to migrate all the apache crap off here and it will be fine |
| 02:08.41 | Ariel_ | otaku_p0pe, I have seen a wrt54g running asterisk |
| 02:08.42 | otaku_p0pe | occasioally you get pop's and stuff |
| 02:08.47 | otaku_p0pe | Ariel_: hott |
| 02:08.57 | nick125 | whats a good, free, low b/w codec for asterisk? |
| 02:09.04 | wulfy814 | Ariel_ : ok, hints on how to make asterisk start automagically in ubuntu then? |
| 02:09.04 | Ariel_ | gsm |
| 02:09.13 | Ariel_ | g726-32 |
| 02:09.17 | nick125 | that would be able to support least 5 calls at once on a 1mb up/down line |
| 02:09.21 | wulfy814 | BrainSurg: you just needed to do an "apt-get install" right? |
| 02:09.27 | Ariel_ | ilbc works also but has heavy cpu load |
| 02:09.41 | BrainSurg | wulfy814: yeah |
| 02:09.43 | adelas | Ariel_ will you be my best asterisk friend :P |
| 02:09.44 | Ariel_ | nick125, 5 calls ulaw would work |
| 02:10.06 | Ariel_ | adelas, ???? |
| 02:10.09 | BrainSurg | wulfy814: Autostarts, too. |
| 02:10.22 | adelas | need amp help :P |
| 02:10.57 | wulfy814 | BrainSurg: zaptel hw or ztdummy? |
| 02:11.17 | Ariel_ | adelas, what is your issue with amp? |
| 02:11.25 | *** join/#asterisk Mycroft1 (n=reece@whatda.whatthe.net) |
| 02:11.31 | Mycroft1 | hi pplz :) |
| 02:12.20 | adelas | everything ;) |
| 02:12.34 | adelas | the AMP installation guide is way too outdated :| |
| 02:12.36 | adelas | as it seems heh |
| 02:12.55 | Ariel_ | do you have a box just for the asterisk? |
| 02:13.21 | adelas | yes |
| 02:13.44 | Ariel_ | then get yourself a major time saver and put Asterisk@home on it. |
| 02:13.47 | adelas | i'v beeen jumping back from fedora with asterisk to asteriskAtHome, then asteriskwin32 |
| 02:13.55 | adelas | then now i'm back to fedora :| |
| 02:14.09 | adelas | i have asterisk running |
| 02:14.17 | adelas | just now the amp stuff |
| 02:14.23 | Ariel_ | fedora is beta from RH CentOS is actual RHEL 3 |
| 02:14.37 | Ariel_ | adelas, amp needs lots of other things added. |
| 02:14.40 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 02:14.43 | adelas | yea thats the thing :| |
| 02:14.58 | adelas | i did all that "yum install package" stuff |
| 02:15.19 | adelas | followed the steps, then bam it skips things and tells to do things out of nowhere |
| 02:15.30 | Ariel_ | adelas, did you install asterisk::perl, and all the others as well? |
| 02:15.42 | adelas | yea |
| 02:15.54 | adelas | wait, u mean perl, and perl-CPAN |
| 02:16.08 | adelas | or a acutally "asterisl::perl" |
| 02:16.35 | adelas | o wait, i did do like perl -MCPAN -e "install IPC:Siglal" |
| 02:16.38 | adelas | nand stuff like that |
| 02:16.41 | adelas | and* |
| 02:17.04 | pbd | Ok, I'm a cvs dummy.. can anyone tell me how to check out the Makefile for -HEAD from Digium? |
| 02:17.52 | Ariel_ | pbd, it comes with asterisk when you do the co asterisk |
| 02:18.00 | *** join/#asterisk ChArLeS___ (n=charles@64.35.168.55) |
| 02:18.03 | ChArLeS___ | wow |
| 02:18.09 | pbd | Ariel_: Not my question, but thanks anyway. |
| 02:18.53 | adelas | Ariel_ could u help me with AMP? |
| 02:19.01 | ChArLeS___ | Hey |
| 02:19.05 | adelas | like how to get started heh |
| 02:19.12 | ChArLeS___ | Why asterisk forks for each IAX call ? is it normal ? |
| 02:19.48 | pbd | My -HEAD makefile became corrupted, and I wanted to just get a new one. |
| 02:20.07 | pbd | Fortunately, I have more than one server with -HEAD on it, so I grabbed it from there. but.. eeww. |
| 02:21.57 | otaku_p0pe | victory ! |
| 02:24.04 | hardwire | hmm |
| 02:24.05 | hardwire | so |
| 02:24.06 | hardwire | iaxy |
| 02:24.07 | hardwire | I have one |
| 02:24.19 | hardwire | I am hoping to make it more or less auto dial when the line is picked up |
| 02:24.57 | pbd | Can you also use yours as a handy stove/furnace replacement? |
| 02:25.05 | mago3-cn | hello, how can i change the caller id, depending on what it is? |
| 02:26.49 | pbd | Hardwire: I believe the current quote is- your watch has more processing power than an Iaxy. |
| 02:27.02 | hardwire | I wasn't worried about the iaxy :) |
| 02:27.21 | pbd | mago- can you be more specific as to what you're trying to do? |
| 02:27.38 | pbd | Change the caller id based on what WHAT is? |
| 02:27.42 | hardwire | as soon as I pickup the phone it gives Accepted AUTHENTICATED TBD call from 10.10.0.95 |
| 02:27.57 | hardwire | I just want it to patch through on the asterisk side to a given number |
| 02:28.19 | pbd | hardwire- that will depend on the context you have the iaxy dropping into. |
| 02:28.28 | hardwire | local |
| 02:28.38 | hardwire | which has all sip phones |
| 02:28.38 | *** join/#asterisk frank_sbr (n=frank_sb@modemcable175.124-70-69.mc.videotron.ca) |
| 02:28.51 | pbd | Right.. but what does local do on the no-match case? |
| 02:28.51 | frank_sbr | hi |
| 02:29.08 | hardwire | pbd: nothing.. and wouldn't asterisk respond if there were a no match case? |
| 02:29.12 | frank_sbr | does anyone knows how to change ilbc 30ms to 20ms encoding length |
| 02:29.15 | hardwire | like |
| 02:29.18 | hardwire | no match found.. blah. |
| 02:29.32 | hardwire | actually.. what is the extension code for no match found. |
| 02:30.45 | hardwire | just s |
| 02:30.46 | hardwire | or o? |
| 02:31.00 | pbd | s would work.. but what's your iax.conf say? |
| 02:32.01 | hardwire | good enough |
| 02:32.06 | hardwire | pastebin.. slow |
| 02:32.16 | meppl | gute nacht - good night |
| 02:32.53 | hardwire | http://pastebin.com/340417 |
| 02:32.57 | hardwire | .ca was too slow |
| 02:33.16 | hardwire | iaxcompat=yes in general |
| 02:33.23 | hardwire | and just codec stuff after that |
| 02:34.25 | nick125 | hmm |
| 02:34.30 | pbd | I've got an s,1,Answer in my gneral context.. |
| 02:34.40 | nick125 | when i try to dial 777, it says '484 address incomplete' |
| 02:34.42 | pbd | that should catchall everything that winds up there. |
| 02:34.47 | JerJer | MXC is so goddamn funny |
| 02:34.59 | hardwire | pbd: yeh.. but the iaxy is waiting to interperate some dtmf |
| 02:35.20 | hardwire | then it gives me the congestion.. all without talking to asterisk |
| 02:35.24 | pbd | JerJer- since you're awake, can you answer a quick works/noworks question for me for the nufone h323 driver? |
| 02:35.27 | hardwire | after a long dely at dialtone |
| 02:35.48 | JerJer | depends |
| 02:35.49 | pbd | hardwire- ok, so the iaxy isn't configured correctly.. for that, I'd have to have one around to mess with. |
| 02:35.59 | nick125 | anyone know why 777 isnt working? |
| 02:36.09 | JerJer | no luck tonight |
| 02:36.22 | hardwire | pbd: there are no real options for this in the iaxy :) |
| 02:36.38 | pbd | JerJer: Situation is outbound (HEAD to cisco callmanager in this case, via gatekeeper) out of band dtmf. |
| 02:36.55 | JerJer | rfc2833 |
| 02:36.57 | pbd | JerJer: While * is accepting oob dtmf, it doesn't seem to be sending it on. |
| 02:37.09 | pbd | Configured for rfc2833 all around. |
| 02:37.10 | Mycroft1 | has anyone ever seen this error before, i cant find any info on it and i get 400 of these messages on my console per second chan_iax2.c:2455 iax2_read: I should never be called! |
| 02:37.38 | JerJer | don't call it |
| 02:37.40 | pbd | hardwire: have you done an iax show peers? |
| 02:37.56 | hardwire | the iaxy connects to the pbx |
| 02:38.07 | hardwire | I can dial to any regular extension thats in my context |
| 02:38.11 | hardwire | via a regular phone |
| 02:38.16 | hardwire | I just need a bat phone |
| 02:38.24 | Mycroft1 | yeah iax2 show peers is fine |
| 02:38.34 | hardwire | oh. |
| 02:38.34 | hardwire | heh |
| 02:38.37 | hardwire | bleh |
| 02:38.39 | pbd | (ok, so I dropped the 2) |
| 02:38.53 | Mycroft1 | thats cool :) |
| 02:39.07 | Mycroft1 | its been driving me nuts because all the calls start getting audio gaps while it does this |
| 02:39.10 | Mycroft1 | i cant find the problem |
| 02:39.13 | pbd | JerJer: ever heard of the dtmf problem before, or did I dig up something new? |
| 02:39.21 | Mycroft1 | it just started when i upgraded to 1.0.8 |
| 02:39.36 | pbd | (or worse, it's my config- which doesn't seem likely right now, but is generally the case) |
| 02:40.19 | nick125 | now to find some hold music :P |
| 02:40.31 | drray | nick - use a dialtone.mp3 :) |
| 02:40.53 | pbd | Mycroft- I can't say that I've seen that one.. but it could be a bad compile.. you're running .8, downloaded as tarball? |
| 02:40.57 | nick125 | lol |
| 02:42.31 | pbd | There's always Allison's rendition of louie louie.. but I'll bet you want callers to wait on hold, don't you? |
| 02:42.33 | hardwire | whats 'phonecore' |
| 02:42.39 | nick125 | lol |
| 02:42.40 | hardwire | http://asterisk.gnuinter.net/files/digium/phonecore/ |
| 02:43.17 | pbd | It's that little place in the middle of the dial where they put the paper with your number typed on it. |
| 02:44.03 | Mycroft1 | pbd: its running on freebsd so from the ports tree |
| 02:44.28 | pbd | Mycroft: You like punishing yourself, don't you? |
| 02:44.32 | *** join/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg) |
| 02:44.47 | Mycroft1 | pbd: linux does not like me :) |
| 02:45.09 | pbd | Mycroft- I'd almost guarantee that it's some sort of bad compile issue- but I don't run the bsd version, so there's not much else I can say. |
| 02:46.37 | nick125 | i wonder why hold music crashes asterisk :/ |
| 02:46.53 | Mycroft1 | pbd: possable i dont have zaptel running, but everything seems to be working MOH and MeetMe |
| 02:47.21 | Ariel_ | nick125, did you remove the id tags from the mpg files you uploaded to asterisk box? |
| 02:47.26 | hardwire | nick125: its not real music. |
| 02:47.27 | Mycroft1 | then again i couldnt see ztdummy in the make file and since this box doesnt have a digium card it was pointless to install it |
| 02:47.32 | pbd | nick: likely that your zaptel is messed up, and it can't provide adequate timing. |
| 02:47.55 | nick125 | Ariel_: oh that might help |
| 02:48.24 | pbd | Mycroft- the ztdummy stuff is in the zaptel package makefile.. but if you don't need moh or meetme, you're right- who cares. |
| 02:48.41 | *** join/#asterisk jeobjeobjeob (n=jeobjeob@pool-70-111-134-244.nwrk.east.verizon.net) |
| 02:48.43 | jeobjeobjeob | hey |
| 02:48.44 | pbd | Now, it *might* affect other timing stuff, if the kernel is off- but again, I'm no bsd expert. |
| 02:48.54 | jeobjeobjeob | im running into an issue with cdr and dial |
| 02:49.04 | jeobjeobjeob | when i dial another number, is it possible to have the cdr forked? |
| 02:49.07 | jeobjeobjeob | in a way so that |
| 02:49.32 | Mycroft1 | pbd: zaptel timing wouldnt be causing an iax2 issue though would it |
| 02:49.32 | jeobjeobjeob | the source is the system, and the destination is the phone that picks up the call (in the case of a simulring) |
| 02:49.35 | pbd | jeobjeob: Anything is possible, you've got the source and a c compiler, right? ;-) |
| 02:49.52 | Mycroft1 | pbd: i remember back in the old 0.x days it was very touchy |
| 02:50.06 | pbd | Mycroft- No, but asterisk uses zaptel as a timing source, if it can... so if that's off, little gaps could occur. |
| 02:50.06 | jeobjeobjeob | better question is: where does the channel merging occur? |
| 02:50.43 | ChArLeS___ | ps auxwww |grep asterisk |wc |
| 02:50.43 | ChArLeS___ | <PROTECTED> |
| 02:50.49 | pbd | With modern (linux 2.6) kernels, the kernel has a high enough resolution timer to make the ztdummy stuff obsolete. |
| 02:50.51 | ChArLeS___ | why each call is a fork ? |
| 02:51.03 | *** join/#asterisk _santiago_ (n=santiago@63.245.86.202) |
| 02:51.16 | pbd | But you're running bsd- so... no clue. |
| 02:51.23 | Mycroft1 | pbd: might explain the audio gaps sometimes |
| 02:51.43 | Mycroft1 | i might try over the weekend, move some of the calls off to another asterisk box and see what i can do with it :) |
| 02:51.49 | Sedorox | zaptel + bsd = bad voodoo |
| 02:52.07 | Sedorox | tho I never had problems with it |
| 02:52.21 | Mycroft1 | its just a pain all the other asterisk linux to freebsd moves worked seamless except the last one |
| 02:52.22 | pbd | Mycroft- but your asterisk is sick in general, if it's giving you error messages in the iax.c code. |
| 02:52.25 | Mycroft1 | typical :) |
| 02:52.42 | Mycroft1 | i cant explain it |
| 02:52.44 | Sedorox | yea... always the last one |
| 02:52.49 | Mycroft1 | all the other 38 asterisk boxes are fine |
| 02:52.58 | Sedorox | maybe ya got bad memory in it.. and it messed the compile up.. |
| 02:53.08 | Mycroft1 | maybe |
| 02:53.13 | pbd | Suggestion? Stick with 38 boxes. :) |
| 02:53.23 | pbd | 39 being a multiplier of 13. |
| 02:53.26 | Mycroft1 | i might roll the ports tree off another one thats working and the working directory |
| 02:53.28 | Sedorox | lol |
| 02:53.29 | Mycroft1 | see if its any better |
| 02:53.49 | pbd | er.. multiple. It's late, time for me to go- I've been beating my head against this dtmf issue too long today. |
| 02:53.55 | Mycroft1 | pbd: this last one is for the office in melbourne |
| 02:54.23 | Mycroft1 | not dtmf arrgg, i still feel the pain from that with the cisco 5300's |
| 02:54.50 | niZon | mmmm message waiting indicator |
| 02:54.50 | pbd | Mycroft- you ran dtmf's to cisco equipment via h.323? |
| 02:55.04 | *** join/#asterisk Ferris_B (n=silentra@ip68-101-182-205.sd.sd.cox.net) |
| 02:55.15 | Mycroft1 | nope sip |
| 02:55.35 | Mycroft1 | we also found that we had echo after we got to 100 calls |
| 02:55.40 | Sedorox | from what I've seen in here... h323 seems like the devil... |
| 02:55.43 | nick125 | would it be better if i put my hold music as wavs ? |
| 02:55.44 | pbd | Mycroft- rats. Sip to cisco for me works like a charm- if only Cisco supported compressed codecs under SIP in callmanager. |
| 02:55.49 | Mycroft1 | we tried new, ios's firmware on the voice cards |
| 02:55.50 | Mycroft1 | everything |
| 02:55.56 | Mycroft1 | only a reboot gets rid of it |
| 02:56.10 | pbd | I've got to make these connections from Brazil to the US- no way in heck I'm trying that with ulaw/alaw. |
| 02:56.34 | JerJer | why with H.323? |
| 02:56.56 | pbd | Callmanager supports g.729 under h.323 trunks. |
| 02:57.16 | JerJer | call munger also support SIP trunks |
| 02:57.23 | Mycroft1 | pbd: do you run into echo on the calls at all ? |
| 02:57.33 | Sedorox | JerJer, [22:55] pbd Mycroft- rats. Sip to cisco for me works like a charm- if only Cisco supported compressed codecs under SIP in callmanager. |
| 02:57.36 | pbd | Supports them yes- but *only* for ulaw/alaw codecs. |
| 02:58.06 | nick125 | wav would mean less stress on server then mp3, right? |
| 02:58.44 | Sedorox | actually I think raw gsm format is less.. |
| 02:58.48 | pbd | Mycroft- No, no echo whatsoever.. but it's still a new app- yet to be stress tested. |
| 02:58.53 | Sedorox | thats how I have my ivr... |
| 02:58.54 | nick125 | for hold music that is |
| 02:59.10 | nick125 | since wav doesnt have to be decoded |
| 02:59.13 | Mycroft1 | pbd: how many calls are you doing at peak |
| 02:59.17 | Sedorox | moh.... *think* I've done mp3's... dunno what the load was tho :p know it did get too high sometimes.. |
| 02:59.24 | Mycroft1 | pbd: we ran nicely at 20 calls without issue |
| 02:59.34 | nick125 | on mine, with one call, it skips and all |
| 02:59.42 | pbd | Mycroft- under the -STABLE branch, I've seen 30-40 concurrent h.323 calls from CCM to Asterisk using the h.323 driver. |
| 02:59.56 | pbd | And they all wind up in meetme, to boot. |
| 03:00.57 | pbd | Now, I can't run calls from * to CCM under -STABLE- they have to originate on CCM. -HEAD fixes that (there's a bunch o bugnotes with my name on them in mantis to mark each step of that way) |
| 03:01.18 | *** join/#asterisk _daver_ (i=daver@entropy.tmok.com) |
| 03:01.49 | pbd | And, until yesterday, I was ready to go- then I just HAD to test it with a little app I've been working with that does some PIN authentication.. and found that the dtmf doesn't pass from * to CCM. |
| 03:01.51 | Mycroft1 | i think we got to 45 - 50 and we started having echo |
| 03:02.13 | Mycroft1 | pbd: no dtmf issues using h.323 |
| 03:02.27 | Mycroft1 | i might... look back into it, ive got 4 of these 5300's sitting in a store room |
| 03:02.33 | pbd | Mycroft- it's possible- although, my users won't get to that many concurrent for quite some time. |
| 03:02.43 | Mycroft1 | we went to all digium cards once echoing came up |
| 03:02.53 | Sedorox | send a 5300 this way :p |
| 03:03.12 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 03:03.26 | Mycroft1 | sedorox: they aren't mine they're my works :) |
| 03:03.51 | Sedorox | darn :/ |
| 03:03.54 | Sedorox | if you wanna sell them :p |
| 03:04.32 | *** join/#asterisk mino (n=mino@217.172.177.60) |
| 03:04.54 | Sedorox | my rommate and I are putting together a small cisco lab for our room.. |
| 03:05.03 | Sedorox | so I'm trying to grab whatever I can :p |
| 03:07.43 | adelas | does asterisk support voice recording? |
| 03:07.46 | pbd | When we've found echo, we've always traced it to a bad device on the other end. |
| 03:08.00 | pbd | Anyway, I'm outa here. Tomorrow is another DTMF day. Ick. |
| 03:08.27 | Mycroft1 | pbd: i put asterisk on some powermac g5's to see how fast they are |
| 03:08.34 | Mycroft1 | compaired to p4's |
| 03:08.40 | Mycroft1 | its quiet interesting |
| 03:08.52 | pbd | I'm guessing I'm gonna have to start adding debugs to the h.323 channel driver again. It acts as if it just never sees the digits- although the pbx channel debug claims it's handing them off to the channel driver. |
| 03:09.22 | pbd | So.. it could be with the pbx core itself, or it could be with the h.323 driver.. or it could be my config. |
| 03:09.35 | *** join/#asterisk shido6 (n=shido6@d57-87-253.home.cgocable.net) |
| 03:09.46 | Mycroft1 | pbd: so stay away from h.323 and cisco for the moment ? :) |
| 03:09.56 | shido6 | LOL |
| 03:10.10 | SwK | Mycroft1: finding anything interesting on your comparisons? |
| 03:10.12 | pbd | For once, I think it's not the cisco side of the equation. Not usually the case. |
| 03:10.13 | Mycroft1 | i might take one home tonight and have a play see if i can get the h.323 stuff to work with a 5300 |
| 03:10.39 | pbd | If you come up with anything, I'll be around the channel for the next week or so. |
| 03:10.43 | Mycroft1 | SwK: a G5 1.8 SP seem to be able to hold less load per volume of calls |
| 03:10.44 | SwK | i would just be happy if I could get t38modem to work witha 5200 |
| 03:10.46 | shido6 | erf |
| 03:10.49 | shido6 | whats wrong ? |
| 03:10.56 | Mycroft1 | SwK: vs a 2.8 ghz p4 |
| 03:11.03 | *** join/#asterisk elvisthedj (n=kris@host-69-144-177-90.bln-mt.client.bresnan.net) |
| 03:11.04 | SwK | Mycroft1: what OS you running on the G5? |
| 03:11.05 | shido6 | 5300 and h323 works - less someone changed and fx0red something |
| 03:11.12 | shido6 | im in on the tail end of the convo tho |
| 03:11.20 | Mycroft1 | SwK: i tried darwin and freebsd |
| 03:11.32 | SwK | Mycroft1: didnt try OS X? |
| 03:11.38 | pbd | shido- make sure you test that dtmf is working both ways through the channel driver.. that's my wall right now. |
| 03:11.41 | Mycroft1 | SwK: darwin is osx :) |
| 03:11.48 | shido6 | bleh |
| 03:11.50 | SwK | darwin != OS X |
| 03:11.51 | pbd | shido- which channel driver are you using? |
| 03:11.57 | shido6 | I smacked into that wall a while ago and fixed it |
| 03:11.58 | SwK | Darwin is a subset of OS X |
| 03:12.06 | shido6 | the Secret is a single line in h323.conf |
| 03:12.08 | elvisthedj | anybody got a tip on echo cancellation on an iaxy2? (zap not involved.. calling out via sip, no echo on my (iaxy2) end, but the other party hears an echo |
| 03:12.11 | shido6 | I use chan_h323.so of course <--- NuFone |
| 03:12.15 | elvisthedj | i've seen the question on the list, but no answer |
| 03:12.22 | Mycroft1 | Darwin reece-andersons-powerbook-g4-15.local 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc |
| 03:12.27 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) [NETSPLIT VICTIM] |
| 03:12.32 | shido6 | if you look closely at the soure |
| 03:12.34 | pbd | shido- and the line was? |
| 03:12.40 | shido6 | you 'll see a line for dtmf |
| 03:12.44 | shido6 | its hidden in the code |
| 03:12.48 | shido6 | let me go look again.... |
| 03:13.14 | pbd | I only saw two dtmf related lines in the source I was looking at today- dtmfcodec and dtmfmode. |
| 03:13.16 | *** join/#asterisk mino (n=mino@217.172.177.60) |
| 03:13.17 | Ferris_B | Evening, all. Anyone here use sixtel? |
| 03:13.24 | JerJer | dtmfcodec ? |
| 03:13.28 | JerJer | that is documented |
| 03:13.41 | SwK | Darwin bunghole 8.2.0 Darwin Kernel Version 8.2.0: Fri Jun 24 17:46:54 PDT 2005; root:xnu-792.2.4.obj~3/RELEASE_PPC Power Macintosh powerpc |
| 03:13.41 | pbd | Codec, to me is irrelevant, although I set it to 101 anyway- but I'm using out of band signalling. |
| 03:13.45 | *** join/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg) [NETSPLIT VICTIM] |
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| 03:14.07 | *** part/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
| 03:14.11 | pbd | dtmfmode I've got set to rfc2833 for the out of band stuff.. I can't use inband. |
| 03:14.12 | *** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
| 03:14.20 | spackle | jerjer: do you have any tips for troubleshooting line noise on a PRI? I have static or crackles on some channels. |
| 03:14.33 | pbd | (I'm using g.729- inband doesn't work with compressed codecs) |
| 03:14.44 | spackle | jerjer: using a quad span digium card. |
| 03:14.48 | *** join/#asterisk xheliox (n=jeff@user-0c6se1v.cable.mindspring.com) |
| 03:14.50 | JerJer | spackle: sharing interrupts? |
| 03:15.08 | pbd | spackle: running on any hardware that has a blue Dell on the front? ;-) |
| 03:15.25 | spackle | I don't think so, It's running on a proliant DL380 G2. |
| 03:15.32 | *** join/#asterisk blitzrage (n=leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:15.39 | *** join/#asterisk jsaunders (i=js@70.70.37.38) |
| 03:15.48 | spackle | Er, I don't think it is sharing interrupts, that was my first thought too. |
| 03:15.49 | shido6 | ; You may also specify on either a per-peer or per-user basis below. |
| 03:15.49 | shido6 | ;dtmfcodec=101 |
| 03:15.51 | shido6 | is the secret |
| 03:15.57 | JerJer | that's not hidden |
| 03:16.06 | shido6 | if u had 6 yagers |
| 03:16.07 | shido6 | it is |
| 03:16.19 | shido6 | h323.conf.sample |
| 03:16.21 | JerJer | ; Default RTP Payload to send RFC2833 DTMF on. This is used to |
| 03:16.21 | JerJer | ; interoperate with broken gateways which cannot successfully |
| 03:16.21 | JerJer | ; negotiate a RFC2833 payload type in the TerminalCapabilitySet. |
| 03:16.26 | pbd | shido- it has to be commented out, too? ;-) |
| 03:16.29 | JerJer | ; You may also specify on either a per-peer or per-user basis below. |
| 03:16.29 | JerJer | ;dtmfcodec=101 |
| 03:16.32 | shido6 | no, pbd :) |
| 03:17.43 | pbd | JerJer- according to the h.323 trace, it negotiates a payload type 46, dtmf payload- and it accepts it from the other side.. but it won't send it. Refuses- no errors, just swallows it. |
| 03:18.14 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 03:18.28 | nick125_lappy | wow |
| 03:18.40 | spackle | nope, the card is alone on interrupt 7 |
| 03:18.40 | Sedorox | niceeeee |
| 03:18.40 | shido6 | excuse me |
| 03:18.51 | pbd | Ahh.. Zotob is doing it's work again, wiping out parts of the net. |
| 03:18.51 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) [NETSPLIT VICTIM] |
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| 03:18.54 | JerJer | pbd: then don't force a dtmfcodec |
| 03:19.25 | twisted | omg |
| 03:19.41 | twisted | that was a friggin tsunami of a split |
| 03:19.41 | Sedorox | lol |
| 03:19.41 | twisted | good thing i had my surfboard out |
| 03:19.48 | Sedorox | not for us it wasn't :p |
| 03:19.48 | Sedorox | for you it musta been tho |
| 03:19.56 | twisted | well |
| 03:19.56 | Sedorox | only dropped about 100 people.. |
| 03:20.04 | twisted | 362 in the channel rigt now |
| 03:20.05 | Sedorox | from this chan... |
| 03:20.05 | twisted | it went down to 102 |
| 03:20.13 | Sedorox | yea... |
| 03:20.14 | adelas | does asterisk support voice recording? |
| 03:20.22 | Sedorox | it want to like 285 or something |
| 03:20.23 | pbd | Adelas: yes. |
| 03:20.27 | pbd | ~doc |
| 03:20.28 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 03:20.28 | twisted | must have been a multi-way split |
| 03:20.33 | Sedorox | ManxPower, its more like "the servers are splittling" (and repear) |
| 03:20.39 | Sedorox | dunny |
| 03:20.40 | pbd | JerJer- removing it has no different effect. |
| 03:20.41 | Sedorox | dunno* |
| 03:21.06 | adelas | doese asterisk also support conferencing? |
| 03:21.25 | ManxPower | adelas: "show application meetme" |
| 03:21.42 | twisted | adelas, nah. we don't need no stinking conferencing :P |
| 03:21.53 | adelas | no, seriously :P? |
| 03:22.02 | twisted | yes |
| 03:22.18 | adelas | yes its supported :)? |
| 03:22.27 | file[laptop] | conferencing, what's that?!? |
| 03:22.30 | pbd | Manx/Adelas: MeetMe application won't show up without a zaptel compiled (won't compile in) |
| 03:22.44 | twisted | i wouldn't say supported, but it's there ;) |
| 03:22.48 | ManxPower | pbd: I know. |
| 03:22.53 | SwK | *yawn8 |
| 03:23.02 | twisted | and it works |
| 03:23.06 | twisted | SwK, stfu |
| 03:23.09 | ManxPower | twisted: last I saw meetme won't even build without zaptel installed. |
| 03:23.09 | SwK | netsplits rule |
| 03:23.22 | twisted | ManxPower, i didn't say it did |
| 03:23.44 | pbd | Now, for the real question- does Asterisk support videoconferencing? |
| 03:23.58 | SwK | hah |
| 03:24.10 | pbd | Answer: Yes, but it isn't supported. ;-) |
| 03:24.11 | file[laptop] | it supports it in another universe |
| 03:24.14 | ManxPower | pbd: HAHA! That's a good one. Here, I have one! Does Asterisk support T.38? HAHAHAH! |
| 03:24.27 | twisted | ManxPower, nono, i have one better |
| 03:24.32 | ManxPower | What other Asterisk jokes are there? |
| 03:24.36 | JerJer | pbd: how are you determining the dtmf doesn't go anywhere? |
| 03:24.38 | twisted | ManxPower, can asterisk effectively replace a 5ess? |
| 03:24.56 | file[laptop] | or: can Asterisk make me the next Vonage? |
| 03:25.00 | ManxPower | twisted: I'd not heard that one before. |
| 03:25.04 | twisted | ManxPower, lol |
| 03:25.09 | *** join/#asterisk gopherspidey (n=webb@dsl-musc-15-031.machlink.com) |
| 03:25.11 | SwK | asterisk can make you the next vonage |
| 03:25.14 | ManxPower | file[laptop]: Of course it can! Just use your DSL! |
| 03:25.18 | pbd | JerJer: Couple of ways- observation- the application on the other end doesn't receive it.. but through h.323 trace 8, and channel debug, I see the event registered, but no pdu going out. |
| 03:25.27 | ChArLeS___ | ManxPower: I got to see that if using g723 each call is a new fork man |
| 03:25.42 | ChArLeS___ | ManxPower: 2.4ghz can barely handdle only 12 calls |
| 03:25.45 | pbd | JerJer: Yet, if the other side of the conversation (same channel) sends a dtmf tone, I see the pdu come in, and the channel debug registers it. |
| 03:26.02 | ManxPower | ChArLeS___: only mentally instable people use G723.1 |
| 03:26.09 | Qwell | 12 calls on 2.4ghz? |
| 03:26.22 | file[laptop] | G723.1 is a verrrrry CPU intensive codec |
| 03:26.27 | ManxPower | Qwell: Well, the ITU codec isn't optimized, and it's a cpu hog. |
| 03:26.38 | pbd | Charles- er.. are you doing massive amounts of transcoding on that doom server, too? |
| 03:26.43 | ManxPower | not to mention being illegal in much of the world. |
| 03:26.47 | SwK | i love getting requests for 723 |
| 03:27.08 | twisted | 723? |
| 03:27.14 | SwK | trasncoding g729 <-> g723.1 is the bestest |
| 03:27.19 | file[laptop] | O.O |
| 03:27.36 | Netgeeks | you never know, he may have paid for a g723 license! thats why he's running a 2.4GHz system! |
| 03:27.55 | pbd | What's a 723 license run these days, $1K/ channel? |
| 03:27.58 | file[laptop] | yeah and my name is Mr. Muffin Man |
| 03:28.04 | Netgeeks | oooh, I see some digium people here. I have a REALLY dumb arse question |
| 03:28.58 | Netgeeks | is a pci express 16 slot backwards compatable so that you can plug a TE card into it? /duck |
| 03:29.38 | JerJer | pbd: submit a bug with massive detail |
| 03:29.47 | ChArLeS___ | We bought G723 not G729 I don't have G729 so I have to use what my boss got. |
| 03:30.30 | pbd | JerJer: That was my thought- but I wanted to make sure I wasn't just wrong in my config somehow- dtmf seems like the world's simplest setup, yet this problem is really odd- Id think someone else would have seen it. |
| 03:30.34 | ChArLeS___ | probably G729 is also CPU intensive |
| 03:30.48 | file[laptop] | you bought G723... yeah... |
| 03:30.52 | pbd | Charles- IMHE, 729 isn't that bad. |
| 03:31.25 | ChArLeS___ | I can handle more than 60 calls on G711 on that |
| 03:31.27 | ChArLeS___ | box |
| 03:31.35 | file[laptop] | the interweb is crashing! oh no |
| 03:31.41 | ChArLeS___ | file[laptop]: what's the matter ? |
| 03:31.51 | JerJer | pbd: however, I did a quick test on the H.323 load for 7960s and DTMF worked as expected |
| 03:32.01 | pbd | Sure, 711 has no compression at all- if you're not transcoding it, you should be able to support a *lot* of calls. |
| 03:32.07 | file[laptop] | Asia is really sucky right now for routing |
| 03:32.08 | *** join/#asterisk nick125 (n=nick125@unaffiliated/nick125) |
| 03:32.26 | file[laptop] | 52% average packet loss, 670ms average response time |
| 03:32.27 | *** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfk24.dialup.mindspring.com) |
| 03:32.38 | ManxPower | I'm a fan of G726 and Speex |
| 03:32.42 | file[laptop] | lots of routers are down there it seems :( |
| 03:33.01 | pbd | JerJer- I haven't played with the h.323 load for a 7960, I'm running * with 7960's attached via SIP. If my channel is connected to CCM via SIP (testing, I can't run that in production), dtmf works fine. |
| 03:33.31 | ManxPower | pbd: can you make your cisco send payload 101? |
| 03:33.36 | pbd | But, doesn't matter if I use the 7960, or an IAX-attached softphone, outbound * via h.323 doesn't wanna talk oob dtmf. |
| 03:33.40 | ChArLeS___ | I'm making a SIP -> IAX ->IAX -> SIP gateway |
| 03:33.53 | Qwell | ChArLeS___: why? |
| 03:34.28 | Juggie | if anyone uses agi extensively, could you please chime in on bug# 4854 on the bug tracker.... plz tnx. |
| 03:34.30 | pbd | Tomorrow, I'll pull together the relevant traces and submit it. Hard to show- there's nothing in the logs to show that it's failing- it just doesn't happen. If there were error messages, I'd be happier. |
| 03:34.31 | *** join/#asterisk oplog2 (n=oplog2@206.222.29.50) |
| 03:34.52 | ManxPower | Juggie: Honestly I don't think Asterisk should work around the oddities of every language out there. |
| 03:35.19 | ChArLeS___ | Qwell: to save bandwith |
| 03:35.31 | pbd | Manx: I'd have to research that one. I will say that, if I call out from the 7960 through a Sipura to PSTN, then back in to my CCM box, all works well there, too. |
| 03:36.22 | pbd | Manx: the 7960's dtmf options are differently defined- the choices are inband and 'avt' - avt works for rfc2833. |
| 03:36.36 | Juggie | ManxPower, it still isnt reversed in stable...and even so, on head i dont feel a dont send sighup agi flag would be unreasonable. |
| 03:36.50 | Juggie | the suggestion that i should load perl to disable sighup, and then run php inside is lame i wont do that |
| 03:37.02 | Juggie | i would do a custom compile of php to enable thread control first. |
| 03:37.10 | pbd | Net effect is, my testing has narrowed it down to use of the h.323 channel driver, with a working call, will receive but not send oob dtmf via rfc2833. |
| 03:37.14 | *** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 03:37.28 | ManxPower | Juggie: I would prefer something that allowed you to specify which signals to send via SetVar before calling AGI |
| 03:37.37 | pbd | Which sounds like a nice, clean bug.. if there ever is such a thing. |
| 03:38.08 | Juggie | manx, that would work for me too, i dont really care how i can enable/disable it, as long as it can be done via dialplan logic. |
| 03:38.24 | Hogie | I threw away 250k lines of bleh today, and starting over from scratch and have to the 9th to build it again. I dont think I'll do it |
| 03:39.22 | pbd | Hogie- sounds like you need more monkeys and typewriters. |
| 03:39.51 | Hogie | that's actually kinda funny, since I had them change my business cards to "Network Monkey Boy" |
| 03:40.10 | Hogie | got 500 of them printed that way, and the president of the company was pissed that I did that |
| 03:40.26 | pbd | Why, did he want that on his card, too? |
| 03:41.20 | Hogie | she's anal... I was in blue jean shorts today... (I flew this morning before going into work), and before I could change, I got swamped with "this is broke, that is broke", so when lunch came around, I was like, Im not changing clothes, I dont wanna wash something for tomorrow" and she was all mad |
| 03:41.42 | adelas | do i have to setup a trunk in order to get my cisco phones to work? |
| 03:41.47 | twisted | Hogie, that's a lot of bleh |
| 03:41.58 | adelas | b/c i just wanna see if i get my cisco phones to call each other |
| 03:42.55 | Hogie | twisted: you have no idea... I had a nice start on the project, handed it off to someone else, and they totally fucked it up, spent the last 2 weeks trying to fix it to make it work for now, but tonight I just scrapped the whole thing and am going to start over |
| 03:43.30 | pc4 | Hogie - Blue jean shorts? That is pretty bad though =) |
| 03:43.42 | twisted | Hogie, yeh... that sucks |
| 03:43.43 | pc4 | Blue jeans are -ok- here, but I don't know how well those shorts would pass. |
| 03:43.50 | pc4 | HEhe -- but I can get away with flip flops =) |
| 03:44.11 | pbd | G'night all. |
| 03:44.14 | Hogie | I wear jeans everyday except when I know someone is coming to the office that needs to be impressed, or I go out to a meeting |
| 03:44.37 | twisted | Hogie, sounds like my attire |
| 03:44.37 | Hogie | but, when I go flying, I ALWAYS wear shorts right now, we have a heat advisory where I live today (and tomorrow) |
| 03:44.40 | twisted | jeans and a tshirt |
| 03:45.08 | Hogie | I put on a polo shirt or a collared shirt |
| 03:45.14 | *** join/#asterisk jkk_ (i=joe@67.14.192.58) |
| 03:45.16 | twisted | oh, well aren't you fancy |
| 03:45.20 | Hogie | but dockers dont last with me, bluejeans do |
| 03:45.22 | Juggie | hogie, did you ever hang out on efnet? |
| 03:45.26 | Hogie | http://gallery.cyberjunky.net/August18Solo |
| 03:45.29 | Hogie | Juggie: yup |
| 03:45.38 | Hogie | there right now infact |
| 03:45.39 | Juggie | your nick is famaliar |
| 03:45.48 | Juggie | dings a bell |
| 03:45.49 | Juggie | not sure why |
| 03:46.10 | pc4 | Camera on your solo? :P |
| 03:46.19 | Hogie | pc4: dont tell my instructor |
| 03:46.41 | pc4 | tsk tsk... |
| 03:46.42 | pc4 | hehe |
| 03:46.50 | Hogie | http://gallery.cyberjunky.net/August18Solo/P0000460 |
| 03:46.54 | Hogie | I fly models there sometimes |
| 03:46.56 | pc4 | It's hard enough to fly the damn thing as a low time pilot... let alone photos. |
| 03:47.14 | Hogie | pc4: Im the record setter |
| 03:47.15 | Hogie | ;) |
| 03:47.16 | pc4 | hehe |
| 03:47.19 | pc4 | Damn lucky flatlander :P |
| 03:47.27 | twisted | Hogie, hah |
| 03:47.29 | Hogie | once I get up out of traffic |
| 03:47.41 | Hogie | (ie, away from KGPM), I set the autopilot |
| 03:47.43 | Hogie | and take pics |
| 03:47.49 | Hogie | while heading to the practice area |
| 03:47.50 | pc4 | I have to deal with 1/2 mile wide valleys at 7,500 feet densite altitude and 12,500 mountains to the west and north. Basically, no way out :P. |
| 03:47.53 | *** join/#asterisk r0m (n=SysOp@a81-84-68-51.cpe.netcabo.pt) |
| 03:47.59 | twisted | i love it when autopilot flys you into mountains |
| 03:48.00 | pc4 | autopilot on a trainer? :P |
| 03:48.11 | Hogie | autopilot on a c172SP |
| 03:48.15 | Juggie | hogie, you didnt do anything bbs wise, or hang in #celeron #woe anything like that did you? |
| 03:48.16 | pc4 | nice plane :P |
| 03:48.19 | pc4 | 180 horses if its sp |
| 03:48.23 | Hogie | oh yeah |
| 03:48.25 | Hogie | #bbs, #mysticbbs |
| 03:48.30 | Juggie | bingo |
| 03:48.33 | pc4 | 160 horsse at 8,000 foot densite altitue sucks. |
| 03:48.34 | Juggie | iniquity? |
| 03:48.41 | Hogie | yeah, back in the day |
| 03:48.42 | pc4 | You gain like... 300 feet per minute. |
| 03:48.48 | twisted | renegade. |
| 03:48.49 | twisted | bitches. |
| 03:48.55 | Juggie | thats where, iniquity scene :) |
| 03:49.06 | Juggie | renegade was for chumps |
| 03:49.09 | twisted | HAH |
| 03:49.12 | Juggie | iniquity had no hard coded strings :) |
| 03:49.18 | Juggie | not one ! :) |
| 03:49.20 | twisted | neither did renegade |
| 03:49.22 | Hogie | Redneck Technology was my bbs |
| 03:49.24 | twisted | ninny |
| 03:49.30 | Juggie | renegade did |
| 03:49.34 | Juggie | because i remember people would hexedit |
| 03:49.37 | Juggie | to change them |
| 03:49.40 | twisted | hah |
| 03:49.54 | twisted | i never hexedited, and had no stock strings |
| 03:50.06 | Juggie | all i know is iniquity made renegade its bitch. |
| 03:50.11 | twisted | hahahaha |
| 03:50.15 | *** join/#asterisk Grubs (n=Miranda@c220-239-96-230.eburwd5.vic.optusnet.com.au) |
| 03:50.17 | twisted | if you would like to think that, then be my guest |
| 03:50.20 | pc4 | Hogie - What do they charge/hour for that plane, wet? |
| 03:50.22 | Juggie | :) |
| 03:50.33 | *** join/#asterisk ryandude (n=sprakens@203.131.164.162) |
| 03:50.34 | Hogie | $101/hr I think, I could check the site if you want |
| 03:50.36 | twisted | i should setup a telnet bbs |
| 03:50.37 | Juggie | i ran iniquity cdn fidonet hq :) |
| 03:50.41 | twisted | just for shits and giggles |
| 03:50.45 | ryandude | hello everyone |
| 03:50.47 | Juggie | lets play BRE :) |
| 03:50.49 | Hogie | I was suppose to be in one today without an A/P |
| 03:50.51 | twisted | OOH YES |
| 03:50.57 | twisted | Juggie, set up a telnet bbs on your side too |
| 03:50.58 | pc4 | Just wondering... not bad for a 98+ |
| 03:51.00 | Juggie | bre/sre rock. |
| 03:51.01 | twisted | and we'll BRE that shit |
| 03:51.07 | Juggie | hmmmm |
| 03:51.08 | Hogie | omg, BRE! |
| 03:51.11 | Juggie | what was the front end mailer called. |
| 03:51.15 | Hogie | I still have my irex |
| 03:51.16 | twisted | frontdoor? |
| 03:51.19 | Hogie | internet rex |
| 03:51.19 | pc4 | It's a leaseback =) |
| 03:51.21 | Hogie | key |
| 03:51.21 | Juggie | yah! :) |
| 03:51.27 | Hogie | and frontdoor sysop edition |
| 03:51.28 | Juggie | ol frontdoor, taking mail calls. |
| 03:51.28 | Hogie | :) |
| 03:51.32 | twisted | yep |
| 03:51.34 | Juggie | usurper |
| 03:51.35 | Juggie | lord |
| 03:51.36 | *** join/#asterisk bmg505 (n=leon@rndf-146-53-60.telkomadsl.co.za) |
| 03:51.37 | Juggie | lord2 |
| 03:51.38 | twisted | hahaha |
| 03:51.42 | pc4 | MajorMUD! |
| 03:51.42 | Juggie | usurper was cool |
| 03:51.43 | pc4 | =) |
| 03:51.44 | twisted | i had a big ass lord board in nashville |
| 03:51.54 | Juggie | it was the text version of grand theft auto :) |
| 03:51.57 | pc4 | game connection + majorbbs + doom2 was awesome back in the day. |
| 03:52.00 | pc4 | hehe |
| 03:52.03 | JerJer | WWIV |
| 03:52.05 | Hogie | dwango |
| 03:52.07 | twisted | WWIV |
| 03:52.08 | twisted | OMFG |
| 03:52.10 | pc4 | dwango cost too much. |
| 03:52.13 | twisted | there were SO many holes in WWIV |
| 03:52.14 | Juggie | WWIV, didnt like that |
| 03:52.20 | Juggie | iniquity was sexy |
| 03:52.33 | JerJer | yeah it was fun hackin the hell out of wwiv bbs' back in the day |
| 03:52.33 | Juggie | i ran a board called "the underground" for 3 years |
| 03:52.35 | twisted | iniquity was about as sexy as michael jackson in a corset |
| 03:52.40 | tzanger | wwiv!! |
| 03:52.48 | tzanger | I started with telegard and then renegade |
| 03:52.51 | tzanger | but grabbed the wwiv source |
| 03:52.59 | twisted | tzanger, i ran renegade... it rocked |
| 03:53.01 | Juggie | oh yah telegard |
| 03:53.09 | Juggie | a friend of mine swore by that |
| 03:53.09 | twisted | tzanger, i wrote quite a few complex menu systems for renegade |
| 03:53.13 | Juggie | i swore by iniquity |
| 03:53.16 | JerJer | then Wildcat |
| 03:53.17 | Hogie | pc4: http://www.aviatorair.com/graphics/rent/n2128s.jpg |
| 03:53.20 | Hogie | that's what I was flying today |
| 03:53.22 | twisted | tzanger, last i saw, they wound up all over the country |
| 03:53.35 | Hogie | like, the EXACT one |
| 03:53.36 | Hogie | :P |
| 03:53.49 | tzanger | :-) |
| 03:53.51 | twisted | hogie, flying yourself to astricon? :p |
| 03:54.13 | Juggie | http://bbslist.textfiles.com/709/ |
| 03:54.14 | Hogie | http://www.aviatorair.com/graphics/rent/n8963v.jpg is what I usually fly, but the beacon light wasn't working when I preflighted, so they moved me up from it to 1BS |
| 03:54.18 | Juggie | i'm listed :) |
| 03:54.21 | JerJer | avgas is too goddamned expensive |
| 03:54.28 | Hogie | JerJer: $3/gallon... |
| 03:54.34 | JerJer | oh i know |
| 03:54.41 | Hogie | compared to $2.80/gallon for car gas |
| 03:54.42 | Hogie | hmmm |
| 03:54.47 | JerJer | i've gota C172RG |
| 03:54.50 | tzanger | http://bbslist.textfiles.com/519/ was my hood |
| 03:54.52 | tzanger | ice-9 |
| 03:54.59 | tzanger | new gold dream |
| 03:55.27 | Hogie | twisted: I would, but im only at 17 hours flight time.... |
| 03:55.32 | JerJer | $2.51 for regular gas here, today |
| 03:55.33 | *** join/#asterisk HellAgony (i=HellAgon@200.121.236.192) |
| 03:55.42 | tzanger | 519-888-0085 |
| 03:55.43 | tzanger | Waterloo, ON Ice-Nine |
| 03:55.44 | tzanger | (1990) |
| 03:55.50 | Hogie | I just paid $2.89/gallon to fill up my civic |
| 03:56.04 | *** join/#asterisk ZX81 (n=ZX81@222-153-118-124.jetstream.xtra.co.nz) |
| 03:56.11 | JerJer | i want a hydrogen fuel cell |
| 03:56.23 | ZX81 | as a UPS? |
| 03:56.37 | JerJer | as a car |
| 03:56.43 | ZX81 | <PROTECTED> |
| 03:56.45 | Juggie | it wasnt easy to convince my parents to get an extra phone line |
| 03:56.53 | ZX81 | <PROTECTED> |
| 03:57.05 | ZX81 | <PROTECTED> |
| 03:57.06 | JerJer | Juggie: i talked them in to 2 extra phone lines :) |
| 03:57.06 | ZX81 | <PROTECTED> |
| 03:57.11 | Juggie | haha |
| 03:57.13 | JerJer | bling bling baby |
| 03:57.13 | drray | zx81 - just sit on it and go look at the AMPS! |
| 03:57.17 | Juggie | i used to get donations |
| 03:57.23 | Juggie | one time someone decided instead of donations |
| 03:57.31 | Juggie | they would steal road pylons and put them on my driveway |
| 03:57.38 | nick125 | what is a good kbps for the on hold music? |
| 03:57.39 | Juggie | they did that like 3-4 times over a summer |
| 03:57.40 | shido6 | wow - I can only imagine the street mechanics working on that, cigarette in mouth dropping ash in the midst of hydrogen fumes |
| 03:58.08 | brc_ | feel the beat |
| 03:58.10 | brc_ | can ya |
| 03:58.12 | brc_ | can ya feel |
| 03:58.13 | nick125 | 64kbps good? |
| 03:58.19 | brc_ | no |
| 03:58.28 | brc_ | what code |
| 03:58.41 | brc_ | transcode it to native |
| 03:58.46 | nick125 | for the hold mp3 music |
| 03:58.47 | brc_ | so you won't have to on the fly |
| 03:58.49 | shido6 | u know the ones... "hows it running..." then click click... udner the hood... messin with my damn throttle .... "sounds like its a little lean!" revs some more... ( bitch get the fsck out from under my hood! ) |
| 03:58.58 | JerJer | lol |
| 03:59.15 | shido6 | guess its just a detroit thing... but u know what im talking about... |
| 03:59.22 | Hogie | JerJer: my instructor quit on me today |
| 03:59.23 | shido6 | $20 mechanics |
| 03:59.32 | JerJer | Hogie: i would too |
| 03:59.42 | Hogie | something about soloing at 11 hours:( |
| 04:00.18 | Hogie | he kept trying to push me toward the commercial pilot route... |
| 04:00.19 | pc4 | Hogie - quit? lol -- or moved on? |
| 04:00.21 | pc4 | Hehe. |
| 04:00.36 | Hogie | he got a job at United |
| 04:00.38 | Hogie | or something |
| 04:00.40 | pc4 | Hogie - Many instructors are shortimers -- its the way the industry is. They want to move on to commercial. |
| 04:00.42 | pc4 | Yup :P |
| 04:00.43 | JerJer | i don't remember how many hours i had when i soloed |
| 04:00.44 | Hogie | I'd have to read the news again |
| 04:00.46 | hellop | Anyone know how to tell if a Zyxel 2000w is a v.2? Does that just mean it supports SIP v2? |
| 04:00.49 | pc4 | So what do you do for a living? hehe. |
| 04:00.56 | pc4 | Work at an isp? |
| 04:00.59 | Hogie | Im a network Monkey Boy |
| 04:01.06 | Hogie | na |
| 04:01.09 | JerJer | i've got like 1200+ hours now |
| 04:01.13 | nick125 | anyone here know of a good ata device for asterisk (i might get one some day)? |
| 04:01.19 | Hogie | damn, I want my own plane |
| 04:01.37 | loud | buy ariel's plane. |
| 04:01.44 | Hogie | there's a runway by one of our offices that's 1.2 hours driving away |
| 04:01.48 | Hogie | or about 20 minutes flying |
| 04:02.55 | *** join/#asterisk dcm_ (n=Craziman@208.3.11.172) |
| 04:02.56 | Hogie | (mainly cause we have to go west 20 miles on 30mph roads, then south 70ish miles on a 70mph highway) |
| 04:02.59 | JerJer | there is two grass strips within 2 miles of my bat cave |
| 04:03.15 | pc4 | There is a huge runway about 800 feet from my house. |
| 04:03.25 | pc4 | Yes, the jets wake me up at 7 am when it opens =) |
| 04:03.36 | *** join/#asterisk WilliamK (n=wkeller@c-67-172-202-228.hsd1.tx.comcast.net) |
| 04:03.38 | *** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 04:03.39 | Hogie | loud: how much? I'll give $10 |
| 04:03.41 | JerJer | then there is a real airport (uncontrolled) like 11 miles away, which is where we have it hangered |
| 04:04.00 | loud | 10k, his plane is around 45k. |
| 04:04.10 | pc4 | JerJer - Ours is controlled, thankfully. Otherwise it'd probably be suicide -- non standard traffic pattern... one way in... one way out... |
| 04:04.11 | Hogie | what is it? experimental? |
| 04:04.16 | pc4 | 1 mile wide valley :) |
| 04:04.42 | pc4 | And yes, they put damn jets down in it |
| 04:04.43 | pc4 | hehe |
| 04:04.57 | pc4 | The president of flying j slammed his into the mountain next to it a year ago. |
| 04:05.04 | JerJer | sweet |
| 04:05.06 | nick125 | (kinda off topic): is there a way to turn a mp3 into a 64kbps mp3 using transcode or something like that? |
| 04:05.09 | pc4 | Two years before that it was the oracle dude with one of his surplus military planes. |
| 04:05.18 | nick125 | or maybe i could try wav again |
| 04:05.24 | Hogie | KGPM is only 4000ft long, so we dont usually get larger things in |
| 04:05.37 | Hogie | but, we have all the news heli's based there, etc |
| 04:05.45 | pc4 | Well I'm in a resort town, so you get big stuff in that's not meant to fit :P |
| 04:05.50 | JerJer | pc4: i went thru my private at a class c airport |
| 04:06.23 | JerJer | then commerical and everything else was at a 141 school at a class d (atc, no tracon :) |
| 04:06.28 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:06.46 | pc4 | JerJer - You a commercial pilot? Or just get it for the hell of it? |
| 04:07.03 | JerJer | i'm also a CFI |
| 04:07.05 | Hogie | what's the main diff between 141 and the other training? |
| 04:07.06 | pc4 | JerJer - I love flying... the safety aspect just bugs the hell out of me :P |
| 04:07.46 | drray | it's just an unforgiving hobby |
| 04:07.54 | adelas | hey, which sip should i be using for makeing a extention with cisco phones? SIP rfc2833, inband, or info |
| 04:07.56 | JerJer | pc4: i wrote an essay in high school about the future of aviation and won 5th place which was the flight fees for any 141 school in Michigan |
| 04:08.20 | infinity1 | adelas: sounds like you're talking about dtmf. |
| 04:08.22 | pc4 | JerJer - That's not a bad deal. |
| 04:08.25 | infinity1 | adelas: usually rfc2833 |
| 04:08.26 | JerJer | nope |
| 04:08.28 | pc4 | I'm paying for it our of my paycheck... and it HURTS. |
| 04:08.37 | pc4 | JerJer- This is where I'm at: http://www.airnav.com/airport/SUN |
| 04:08.44 | adelas | k thx infinity1 |
| 04:08.49 | Grubs | Dare I ask an asterisk Q? - which router/firewall distros besides ClarkConnect do people install asterisk onto? |
| 04:08.53 | adelas | wats dtmf? |
| 04:09.05 | *** part/#asterisk santiago (n=santiago@63.245.86.202) |
| 04:09.10 | *** part/#asterisk dcm_ (n=Craziman@208.3.11.172) |
| 04:09.15 | JerJer | http://www.airnav.com/airport/KFNT <-- that's where i did my private |
| 04:09.44 | pc4 | Why is it that I'd rather learn my private at a non land-locked valley-surrounded high elevation airport? |
| 04:09.46 | *** join/#asterisk Beccara (n=Beccara@219.89.209.122) |
| 04:09.52 | JerJer | http://www.airnav.com/airport/KTVC <- that's where i did my instrument, commercial, multi-engine and CFI |
| 04:09.55 | pc4 | I suppose I'll become a better pilot for doing it here :P |
| 04:09.55 | Grubs | Is IPCop + asterisk on the same box a possibility? What about monowall? |
| 04:10.01 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:10.10 | pc4 | Do you think? |
| 04:10.14 | drray | monowall is BSD variant |
| 04:10.23 | nick125 | Grubs: monowall is way too small, it would take a while to get anything near asterisk on there |
| 04:10.29 | pc4 | That's pretty (for Michigan :P) |
| 04:10.36 | nick125 | it doesnt even have perl iirc |
| 04:10.36 | drray | I've run firestarter on an asterisk box |
| 04:10.39 | Grubs | I see |
| 04:10.40 | Hogie | JerJer: honestly, what's the diff between the 2 diff private pilot trainings? Ive been up to long to think of the one besides 141 |
| 04:10.46 | nick125 | nor a compiler |
| 04:11.02 | JerJer | part 141 is highly structured |
| 04:11.05 | pc4 | Hogie - google it. But basically 61 = at your own pace, unstructured. 141 = highly structured curricliam and checks and balances. |
| 04:11.12 | pc4 | You can graduate with 5 hours less, too. |
| 04:11.13 | pc4 | or so. |
| 04:11.16 | Grubs | Thx I know people have ClarkConnect and asterisk on the same box. - I guess its bigger :) |
| 04:11.21 | pygrammer | Listenin' to Zeppelin through the phone :-P finally got ztdummy to work... |
| 04:11.21 | Hogie | 141 is less hours, right? |
| 04:11.27 | pc4 | Hogie - Yes... if you can do it in less. |
| 04:11.29 | JerJer | yes, because you are checked more often |
| 04:11.36 | pc4 | Hogie - It's not much less, but you can use simulator for some of it. |
| 04:11.41 | JerJer | like you do a check ride after each phase |
| 04:11.44 | Hogie | I know im part 141 |
| 04:11.49 | Hogie | Ive done 2 checkrides so far |
| 04:11.50 | pc4 | It really doesn't matter anyways, you probably can't get it in 35 regardless. |
| 04:12.05 | JerJer | instead of just at your private, or commercial or instrument |
| 04:12.13 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:12.22 | adelas | can someone tell me wat this means "/var/www/html/panel/safe_opserver: line 5: 1859 Terminated ./opserver.pl" |
| 04:12.28 | adelas | this is after i added an extention |
| 04:12.30 | heison | ~seen sivana |
| 04:12.31 | jbot | sivana is currently on #asterisk (22h 6m 31s) |
| 04:12.31 | JerJer | Hogie: pop quiz...that's Vx of your aircraft? |
| 04:12.31 | pc4 | I'm a firm believer that it really doesn't matter... one way or the other. |
| 04:12.46 | nick125 | hmm |
| 04:12.55 | Hogie | 64 knots |
| 04:13.03 | *** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co) |
| 04:13.04 | JerJer | ok a tough one... Vne :) |
| 04:13.14 | Hogie | 160 |
| 04:13.24 | JerJer | yay |
| 04:13.27 | Hogie | for my "normal" |
| 04:13.33 | Hogie | Im not sure about the SP's |
| 04:13.35 | Hogie | but the 172M |
| 04:13.37 | Hogie | I do know |
| 04:13.39 | JerJer | yeah |
| 04:13.45 | adelas | can someone tell me wat this means "/var/www/html/panel/safe_opserver: line 5: 1859 Terminated ./opserver.pl" |
| 04:13.57 | blitzrage | nick125: what OS? |
| 04:13.58 | JerJer | adelas: we saw it the first time |
| 04:14.02 | nick125 | blitzrage: linux |
| 04:14.04 | adelas | o hehe :p |
| 04:14.33 | ChArLeS___ | wow |
| 04:14.34 | ChArLeS___ | fuck |
| 04:14.39 | JerJer | Hogie: had any emergencies (real) in flight yet ? |
| 04:14.44 | ChArLeS___ | sip uses the double of the bandwidth |
| 04:15.03 | nick125 | it doesnt show the wav up in aMP either |
| 04:15.04 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 04:15.04 | adelas | so can someone tell me? |
| 04:15.04 | Hogie | just got freaked out today when I pulled the throttle out too fast at the end of downwind... |
| 04:15.08 | nick125 | its chmodded 777 |
| 04:15.10 | Hogie | it did a BANG! sound |
| 04:15.19 | JerJer | Hogie: backfire...hell yeah |
| 04:15.33 | pc4 | JerJer - Have you had any? |
| 04:15.40 | JerJer | lol yeah |
| 04:15.40 | Hogie | I was like "Grand Prairie Tower, Cessna 121 Bravo Sierra reporting base, would like a full stop" |
| 04:15.43 | JerJer | 2 |
| 04:15.48 | pc4 | Do explain. |
| 04:15.50 | Hogie | cause I didn't know what it was |
| 04:15.54 | Hogie | never had it happen before |
| 04:16.13 | Hogie | by the time I landed, I was shaking I was so freaked out, lol |
| 04:16.38 | pc4 | I had a dust storm come on rapid approach from over the mountain. It was scary as crap. Luckily the instructor was there and did one hell of a short field landing. I called ATIS after we landed and 30 seconds later it was 25 kts crosswinds. |
| 04:16.47 | ChArLeS___ | when is cisco going to support IAX ? |
| 04:17.01 | Hogie | well, I guess it was when I got out to tie down the plane I was shaking |
| 04:17.14 | JerJer | first one was pre-solo, so i was with my instructor, of course. We were on climb-out, about 2000 feet when all of the fuses popped |
| 04:17.22 | JerJer | so no electric |
| 04:17.33 | nick125 | mp3 works, but, mp3 skips and all |
| 04:17.46 | pc4 | JerJer - Solution? |
| 04:17.47 | JerJer | we just rolled back into the traffic pattern - shook our wings and got Green lighted by ATC to land |
| 04:18.03 | pc4 | And the second one? |
| 04:18.10 | JerJer | fun |
| 04:18.26 | JerJer | i already had instrument and commercial ratings |
| 04:18.36 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 04:18.47 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 04:19.01 | pc4 | =) |
| 04:19.15 | JerJer | up in Traverse City and my recently acquired girlfriend finaly decided to go flying with me for a nice weekend at a casino in the UP |
| 04:19.33 | pc4 | and subsequently released girlfriend? hehe |
| 04:19.38 | pc4 | Ok... go on |
| 04:19.40 | JerJer | about 30 minutes into the flight i noticed the oil pressure fluxuating |
| 04:19.43 | JerJer | then go to nothing |
| 04:19.51 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 04:19.55 | JerJer | then fire blows out of the cowling |
| 04:19.59 | pc4 | Oil pump went out "ohh shit" -- going to throw a rod soon. |
| 04:20.01 | JerJer | girl friend screeches |
| 04:20.20 | JerJer | i unplug her headset, turn off the gas and magnetos |
| 04:20.38 | pc4 | So you could hear atc over her screams, right:? lol |
| 04:20.54 | pc4 | <JerJer> up in Traverse City and my recently acquired girlfriend finaly decided to go flying with me for a nice weekend at a casino in the UP |
| 04:20.56 | pc4 | <JerJer> about 30 minutes into the flight i noticed the oil pressure fluxuating |
| 04:20.56 | pc4 | <JerJer> then go to nothing |
| 04:21.01 | pc4 | <JerJer> then fire blows out of the cowling |
| 04:21.03 | pc4 | There you go :) |
| 04:21.06 | doughecka | sweet |
| 04:21.14 | doughecka | didja crash? |
| 04:21.17 | JerJer | and get on the radio "Minneapolis center, Cessna 6396 Kilo declaring an emergency fire in engine" |
| 04:21.53 | JerJer | and i spiral down to an open field - gril friend now clutching the seat and still screaming |
| 04:21.59 | doughecka | LOL |
| 04:22.14 | doughecka | organ music! |
| 04:22.22 | pc4 | spiral -- as in rapid descent w/ flaps? |
| 04:22.22 | JerJer | the instant the wheels stop rolling, girl friend exits the plane and falls onto the ground |
| 04:22.33 | pc4 | You landed on the field? =) |
| 04:22.44 | JerJer | pc4: no flaps - we want all the time we can get |
| 04:22.47 | pc4 | You told her to immediately exit to the REAR, right? |
| 04:22.50 | JerJer | but yes a pretty tight spiral |
| 04:23.06 | pc4 | JerJer - Ok -- I thoguht you wanted on the ground ASAP, due to fire. |
| 04:23.09 | *** join/#asterisk Cresl1n (n=Cresl1n@adsl-67-126-59-49.dsl.pltn13.pacbell.net) |
| 04:23.12 | pc4 | I'm guessing the fire was extinguished by then. |
| 04:23.26 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 04:23.34 | pc4 | What did the farmer say? :P |
| 04:23.45 | JerJer | yeah the instant i killed the gas the motor stopped and the relative wind blew out the flames |
| 04:23.46 | pc4 | Do continue |
| 04:23.54 | pc4 | Well that's the good news |
| 04:24.03 | pc4 | Could have been worse. |
| 04:24.06 | JerJer | yeah |
| 04:24.24 | JerJer | then i get back on the radio and give a postion report to minn. center |
| 04:24.37 | JerJer | they dispatch the local fire department |
| 04:24.38 | doughecka | what caused the issue? |
| 04:24.45 | JerJer | and the mechanic from the college |
| 04:24.59 | JerJer | doughecka: oil line burst |
| 04:25.05 | doughecka | ah |
| 04:25.08 | doughecka | user error? |
| 04:25.09 | doughecka | :P |
| 04:25.26 | doughecka | or just component fail :P |
| 04:25.31 | pc4 | Might have been mechanical there. |
| 04:25.34 | JerJer | the mechanic proceeded to replace the oil line and said, "alrighty she's ready to fly again" |
| 04:25.35 | pc4 | hehe. |
| 04:25.42 | pc4 | And you said -- you fly it home, right? |
| 04:25.43 | pc4 | hehe |
| 04:25.48 | doughecka | JerJer: no more oil? |
| 04:25.53 | JerJer | girl friend flips out "there is no way in hell i am getting back into that plane" |
| 04:26.02 | doughecka | no more girlfriend? |
| 04:26.10 | JerJer | so girl friend drives mechanics truck back to traverse city and he and i fly home |
| 04:26.19 | JerJer | home = traverse city |
| 04:26.20 | doughecka | lol |
| 04:26.30 | pc4 | So.... how did the relationship work out? |
| 04:26.37 | JerJer | surprisingly she stuck with me for a while longer |
| 04:26.55 | JerJer | and even got back into a plane with me, like 3-4 months later |
| 04:27.12 | *** join/#asterisk bonez41 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 04:27.22 | doughecka | JerJer: while someone pushed it into the hanger? :P |
| 04:27.46 | *** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) |
| 04:28.03 | JerJer | she kept hinting about the mile high club, but being pilot in command and doing the nasty at the same time usually conflct |
| 04:28.06 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 04:28.16 | pc4 | lol |
| 04:28.19 | pc4 | Should have went for it! |
| 04:28.37 | snewpy | jerjer: yeha, plus you need to be houdini to join the mile high club in a Cessna :) |
| 04:28.49 | pc4 | Damn controls to the plane aren't adjustable and too damn low :P |
| 04:28.50 | pc4 | hehe |
| 04:29.06 | JerJer | yeah i can see the headline "two coeds found naked in aircraft ruins" |
| 04:29.09 | *** join/#asterisk ilustrate (i=user@200.92.29.182) |
| 04:29.11 | snewpy | lol |
| 04:29.19 | doughecka | HAH |
| 04:29.37 | Hogie | autopilot... |
| 04:29.40 | Hogie | backseat... |
| 04:29.56 | pc4 | Hogie - Miss the page on weight and balance? :) |
| 04:29.56 | JerJer | then who's watching for traffic? :) |
| 04:30.07 | Hogie | Mr Auto Pilot, lol |
| 04:30.08 | pc4 | Hogie - I'll watch that plane fall right out of the sky. |
| 04:30.24 | pc4 | JerJer - I read a ntsb report about a guy who reported on a regular basis put on auto pilot and "Set an alarm clock" |
| 04:30.24 | pc4 | lol |
| 04:30.27 | Equinox | How can I see what codecs a call is using? |
| 04:30.28 | pc4 | darwin award. |
| 04:30.33 | pc4 | Apparently he had been doing it for years. |
| 04:30.42 | JerJer | Equinox: show channel <foo> |
| 04:30.54 | JerJer | read and write format |
| 04:30.55 | Hogie | would tower tell me if I hit the tail of the airplane on the runway? |
| 04:31.17 | *** join/#asterisk Beccara_ (n=Beccara@219.89.209.122) |
| 04:31.31 | JerJer | pc4: that's hilarious |
| 04:31.34 | pc4 | If they saw it :P |
| 04:31.47 | pc4 | They'll tell you if your baggage door is unlatched. |
| 04:32.20 | pc4 | JerJer - What city you in? |
| 04:32.26 | doughecka | pc4: should the "door ajar" light be a clue? :P |
| 04:32.32 | *** join/#asterisk BrainSurg (n=paul@69.158.227.13) |
| 04:32.52 | Hogie | I just dont remember the tie down ring under the rudder having a flat end before takeoff like it did after that last landing |
| 04:32.54 | JerJer | pc4: the metropolis of Mecosta, MI |
| 04:33.05 | JerJer | right up the street from the Mecosta International Airport |
| 04:33.18 | *** join/#asterisk blake__ (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 04:33.30 | JerJer | http://www.airnav.com/airport/27C |
| 04:33.40 | blake__ | is there any way to have asterisk dial multiple numbers at the same time, and whoever picks up first gets the call? |
| 04:33.48 | shido6 | yes blake |
| 04:33.49 | JerJer | lol i love it when hey 'estimate' gps corrds |
| 04:33.49 | pc4 | Should fly to idaho :P |
| 04:33.50 | pc4 | hehe |
| 04:33.56 | JerJer | hey ? they |
| 04:34.03 | pc4 | Tiny runway. |
| 04:34.13 | JerJer | and elevation - great! |
| 04:34.28 | JerJer | Surface:Â turf, in poor condition |
| 04:34.32 | JerJer | poor is not the word |
| 04:34.37 | JerJer | piss poor maybe |
| 04:34.37 | doughecka | crappy? |
| 04:34.55 | JerJer | NOTAM: "watch out for old tires" |
| 04:35.00 | doughecka | lol |
| 04:35.01 | JerJer | and trash |
| 04:35.17 | JerJer | yeah its 1022 feet with a serious displaced threshold |
| 04:35.27 | JerJer | so call it 800 foot |
| 04:35.42 | JerJer | Obstructions:Â 45 ft. trees, 360 ft. from runway, 108 ft. left of centerline, 8:1 slope to clear |
| 04:35.50 | pc4 | Is that your "1 aircraft based on field"? |
| 04:35.51 | pc4 | hehe |
| 04:36.27 | JerJer | sweet - they finally added the model aircraft notam |
| 04:36.41 | JerJer | we've been bitching at the faa for years to add that |
| 04:37.08 | pc4 | IS that your 1 plane that you rent? |
| 04:37.10 | pc4 | hehe |
| 04:37.12 | doughecka | I have a friend whos in new guinea, and the closest airport has a crazy runway |
| 04:37.14 | JerJer | lol - i just noticed the one aircraft on field |
| 04:37.16 | JerJer | that's funny |
| 04:37.23 | BrainSurg | Anyone know where to find zaprtc? |
| 04:37.32 | hellop | Dang... wireless VOIP looks like a big PITA |
| 04:37.34 | doughecka | grassfield... its at an angle, so when you take off your going downwill |
| 04:37.45 | doughecka | and when you take off, you go DOWN |
| 04:37.47 | doughecka | to pickup speed |
| 04:37.56 | doughecka | and quickly go hard left |
| 04:38.04 | doughecka | cause the valley turns |
| 04:38.16 | JerJer | sounds like taking off at vegas |
| 04:38.17 | doughecka | at one point your BELOW the runway |
| 04:38.33 | SplasPood | Hrm, does asterisk have any support for "Call Waiting Deluxe" ? |
| 04:38.39 | *** join/#asterisk raphWRKN (n=raph@210.15.240.107) |
| 04:38.40 | JerJer | define deluxe |
| 04:38.44 | doughecka | of course the throttle is at full power till your FAR away :) |
| 04:39.12 | JerJer | nice |
| 04:39.16 | SplasPood | JerJer: Well, there's a service called Call Waiting Deluxe... I've seen some pots handsets which support it... basically when you get the beep it gives you options on how to handle it |
| 04:39.27 | SplasPood | conference, ask calling party to hold, busy... |
| 04:39.29 | doughecka | yea, hes a commercial pilot, but hes only flown out once :) |
| 04:39.35 | SplasPood | forward.. |
| 04:39.47 | JerJer | yeah i like to fly forward :P |
| 04:40.10 | JerJer | doughecka: sounds like fun |
| 04:40.21 | h3x | its called using a voip phone |
| 04:40.24 | JerJer | i flew sky divers for a few months |
| 04:40.31 | doughecka | lol |
| 04:40.39 | h3x | call waiting deluxe that is |
| 04:40.44 | JerJer | i would always try to beat them down |
| 04:40.57 | doughecka | hahaha |
| 04:41.03 | doughecka | as your wings snap off |
| 04:41.16 | SplasPood | h3x: heh, yes.. but I was thinking of when using standard pots phones.. Just curious, I have no application |
| 04:41.24 | JerJer | last one out the door I would do a hammer head and floor the throttle |
| 04:41.29 | SplasPood | setting up some 2.4ghz portable I picked up and I see it in the menu options |
| 04:41.31 | doughecka | hahaha |
| 04:42.03 | *** join/#asterisk BharatS (n=bharatsa@210.211.246.47) |
| 04:42.19 | h3x | SplasPood: well you can sort of do that with adsi phones |
| 04:42.29 | doughecka | man my internet sux |
| 04:42.41 | doughecka | monkey balls |
| 04:42.43 | h3x | but the question is "why" when adsi phones suck and cost more than voip phones |
| 04:42.52 | h3x | if that is what you are looking for |
| 04:42.57 | SplasPood | well thats the thing, I wasn't thinking of ADSI phones |
| 04:43.15 | SplasPood | I was thinking of providing this same type of call waiting service to pots phones |
| 04:43.48 | h3x | by what means |
| 04:43.51 | *** join/#asterisk fad (n=fahad@202.142.189.86) |
| 04:43.52 | pc4 | Well I'm outta here... nice talking guys =) |
| 04:43.56 | JerJer | l8r |
| 04:44.02 | JerJer | i need to get back to the code |
| 04:44.02 | pc4 | Bye =) |
| 04:44.04 | pc4 | hehe |
| 04:44.15 | h3x | if you are talking about with voip adapters for residential users |
| 04:44.21 | h3x | then asterisk has nothing to do with that |
| 04:44.22 | SplasPood | no |
| 04:44.34 | SplasPood | nevermind... |
| 04:44.38 | h3x | well |
| 04:44.48 | h3x | what practical way can you deliver a bunch of lines to people with pots lines |
| 04:44.51 | h3x | that needs this feature |
| 04:45.00 | SplasPood | I'm not talking about delivering service to anyone |
| 04:45.02 | SplasPood | I was just asking |
| 04:45.07 | SplasPood | if I plugged a normal handset |
| 04:45.11 | SplasPood | that supported this feature |
| 04:45.12 | SplasPood | into a card |
| 04:45.13 | SplasPood | in a box |
| 04:45.16 | SplasPood | running asterisk |
| 04:45.17 | SplasPood | zaptel card |
| 04:45.27 | SplasPood | could that type of call waiting indicator/etc be sent. |
| 04:45.56 | h3x | i mean you could hack it |
| 04:46.01 | SplasPood | Since a number of standard (ie, non voip, non adsi, etc) "home" phones seem to have support |
| 04:46.03 | h3x | have dtmf buttons that do things with that call |
| 04:46.13 | SplasPood | it was just a point of curiosity, since I just saw the option for it in a new phone |
| 04:46.17 | h3x | but i cant figure out a practical purpose of it |
| 04:46.17 | SplasPood | true |
| 04:46.30 | h3x | oh that stupid iQ thing? |
| 04:46.37 | h3x | Caller IQ? |
| 04:46.41 | SplasPood | I dunno, it was referred to as "Call Waiting Deluxe" |
| 04:46.44 | SplasPood | but that could be it... |
| 04:46.55 | h3x | Caller IQ is a sick way to get more money out of people that bought a phone |
| 04:46.59 | h3x | its like a lite version of ADSI |
| 04:47.29 | h3x | but i mean |
| 04:47.44 | h3x | i dont see a point in doing call rejection with a call waiting call when you cant do it with a normal incoming call |
| 04:47.47 | h3x | like you can do with a voip phone |
| 04:47.54 | nick125 | it says i have 0 messages, when i call *98, it says i have 5 |
| 04:48.04 | SplasPood | oh yea of course... like I said, just a point of curiosity... |
| 04:48.09 | h3x | maybe your mailbox= string is busted |
| 04:48.28 | BrainSurg | h3x: Dumb question, what is ADSI? |
| 04:48.36 | nick125 | h3x: its set for the same box num |
| 04:48.44 | h3x | I think that CallerIQ stuff is proprietary and isnt a bellcore spec so the precise definition of it wouldnt be implemented in asterisk |
| 04:48.47 | jsmith | Anybody know of a way to set the auto-answer on the Cisco 7960 via the TFTP configs? |
| 04:48.55 | SplasPood | h3x: ahh, that makes sense |
| 04:49.03 | h3x | Analog Display Station Interface or something |
| 04:49.35 | h3x | its a "smart phone" whereby it communicates with the "Telco" using a V.29 FSK 1200bps modem to receive screens and scripts |
| 04:49.42 | h3x | and DTMF or V.29 to transmit data |
| 04:50.00 | h3x | aastra.com has some example adsi phones but now they are mostly going voip |
| 04:50.09 | BrainSurg | h3x: Cool. |
| 04:50.12 | BrainSurg | thanks |
| 04:50.37 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 04:50.45 | Hogie | jsmith: you can't... |
| 04:51.05 | Hogie | that's one of the things listed as not config fileable on cisco's site |
| 04:51.09 | Hogie | for sip firmware |
| 04:51.20 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 04:51.34 | Goshen | Can someone add to the README file the prerequisites that are needed on the system before you make? |
| 04:51.42 | Goshen | like bison, openssl ect... |
| 04:51.52 | Goshen | they are on the asterisk.org webpage, but not in the README file |
| 04:52.35 | Goshen | mpg123 |
| 04:52.47 | jsmith | Goshen: mpg123 is not a *requirement* |
| 04:52.51 | *** join/#asterisk juice (i=1000@mo-67-77-188-19.dyn.sprint-hsd.net) |
| 04:52.55 | Goshen | you get the idea |
| 04:53.00 | jsmith | Yeah, I know... |
| 04:53.09 | *** join/#asterisk spoot_nick (n=julio@50.118.233.220.exetel.com.au) |
| 04:53.20 | Goshen | just need someone with access to take a second and drop a paragraph in there |
| 04:54.12 | *** join/#asterisk immo (n=immo@202.142.189.86) |
| 04:54.45 | ilustrate | can i install asterisk with regular analog phones? |
| 04:55.09 | jsmith | ilustrate: Yes, if you have the right hardware card. |
| 04:55.32 | immo | can I change the sip port for a particular user in sip.conf other than 5060 |
| 04:55.39 | spoot_nick | speaking of which, does anybody know if USA FXS cards would work in Australia? |
| 04:55.49 | BrainSurg | jsmith: Need an FXS interface, right? |
| 04:55.54 | jsmith | BrainSurg: You got it. |
| 04:56.22 | jsmith | spoot_nick: Check the Digium site... I know several of their T1 cards are certified in Australia, but I'm not sure about the TDM series... |
| 04:57.05 | immo | <PROTECTED> |
| 04:57.23 | ilustrate | what would i need to setup a 4 trunk 6 extension setup. i'm trying to do my first commercial install and a client wants to (another client with over 10lines and 6 locations) wants a way to monitor calls before the telefone bills without waiting 2 months. |
| 04:57.41 | Equinox | What do you guys think of the sipura adapters? |
| 04:58.05 | Hogie | my sipura 2100 is making a great paperweight (I haven't even taken it out of the box yet) |
| 04:58.26 | Equinox | I need some sorta analog adapter |
| 04:58.28 | jsmith | ilustrate: Check out CDR. |
| 04:58.33 | Equinox | Just trying to figure out what the least painful one is. |
| 04:58.45 | ilustrate | CDR is that a company or what? |
| 04:59.20 | jsmith | ilustrate: No, it's Call Detail Records.... |
| 04:59.23 | jsmith | ilustrate: Go look them up. |
| 04:59.29 | ilustrate | perfect. o.k. |
| 04:59.47 | jsmith | Equinox: IAXy is the least painful. |
| 05:00.00 | BrainSurg | jsmith: IAXy? |
| 05:00.17 | Equinox | jsmith- Supports iax? Nice. |
| 05:00.19 | jsmith | BrainSurg: Go check Digium's hardware selection... |
| 05:00.24 | jsmith | Equinox: You got it! |
| 05:00.46 | Hogie | oops, I dialed 008 from my home phone (that's hooked up to the work * box), and... I must have just yelled "MUPPET!" into the external loud speaker at work, doh, my bad |
| 05:00.53 | Equinox | jsmith- The S101? |
| 05:01.10 | jsmith | Equinox: Yes, I think that's the one... grey in color, right? |
| 05:01.26 | *** join/#asterisk oej (n=oej@ti111210a080-0810.bb.online.no) |
| 05:01.26 | adelas | hey, i have 2 cisco phones setup that are connected to the phx, one with extention 200 and other with 201 |
| 05:01.30 | adelas | how can i get them to communited |
| 05:01.35 | Equinox | jsmith- Looks black |
| 05:01.35 | adelas | communicate* |
| 05:01.39 | adelas | or call eachtoerh? |
| 05:01.40 | doughecka | holy water |
| 05:01.44 | jsmith | Equinox: Yeah, that's the one... |
| 05:01.55 | adelas | how can i get one of my cisco phones to call the other one? |
| 05:02.14 | spoot_nick | adelas: add an extension for each. check the SIP examples |
| 05:02.19 | jsmith | adelas: Add a dialplan entry, maybe? exten=>101,1,Dial(SIP/102) |
| 05:02.28 | hellop | adelas dial its extension defined in sip.conf |
| 05:02.34 | spoot_nick | and that as well... |
| 05:02.37 | Barmal | why sometimes I keep getting those auto destroying call messages? |
| 05:02.48 | adelas | ok |
| 05:03.05 | BrainSurg | Does anyone know of a canadian mail order place that sells digium hardware? |
| 05:03.58 | adelas | wait, wat entry would i add? |
| 05:04.06 | adelas | this is confusing |
| 05:04.32 | jsmith | adelas: Ok, here goes. |
| 05:04.51 | jsmith | adelas: Let's assume the Cisco phones are setup as users SIP/101 and SIP/102 in sip.conf. |
| 05:04.55 | jsmith | adelas: Are you with me so far? |
| 05:05.08 | Hogie | sip.conf? |
| 05:05.10 | Hogie | what's that? |
| 05:05.16 | ilustrate | are CDR records immediately available? analog? |
| 05:05.21 | adelas | yea |
| 05:05.30 | adelas | well, i have 200 and 201 |
| 05:05.35 | JerJer | define immediately |
| 05:05.35 | jsmith | adelas: OK, that's fine. |
| 05:05.47 | ilustrate | within the hour at least |
| 05:05.48 | Hogie | ilustrate: they are on our system, as soon as the call is taken down |
| 05:05.59 | ilustrate | o.k. |
| 05:05.59 | jsmith | adelas: And in sip.conf, you've set them to start in a certain context, let's call it [internal] |
| 05:06.18 | JerJer | unless you are doing cdr batching |
| 05:06.29 | adelas | okay |
| 05:06.30 | spoot_nick | adelas: as far as I configured mines, asterisk generates CDR csv logs soon as the call is completed |
| 05:06.42 | Hogie | JerJer: you still run nufone, right? |
| 05:06.43 | spoot_nick | adelas: (busy, non answered, answered) |
| 05:06.52 | jsmith | adelas: Then, inside that context (the one I called internal), you'd add an extension. |
| 05:06.54 | spoot_nick | that's the default |
| 05:06.55 | JerJer | nope that was last weeks project |
| 05:07.09 | jsmith | adelas: exten=>200,1,Dial(SIP/200) |
| 05:07.10 | Hogie | heh |
| 05:07.11 | adelas | jsmit, can you pm me with an example to type in? |
| 05:07.17 | jsmith | adelas: exten=>201,1,Dial(SIP/201) |
| 05:07.20 | adelas | ok |
| 05:07.26 | jsmith | adelas: That should do it. |
| 05:07.35 | Hogie | nufone has saved us $162 in the last month |
| 05:07.39 | Hogie | yay 4 nufone |
| 05:07.45 | JerJer | sweet |
| 05:08.09 | jsmith | JerJer: So what are you up to now? |
| 05:08.26 | nick125_lappy | i got a quick question: is it possible for, if after someone is on hold in a certain queue, that it would forward the call to my cell phone? |
| 05:08.54 | adelas | jsmith thx |
| 05:09.09 | adelas | now i gotta figure out how to do multiple phone lines |
| 05:09.14 | jsmith | adelas: No problem... go buy http://www.oreilly.com/catalog/asterisk/ |
| 05:09.29 | BrainSurg | Hogie: What is nufone? |
| 05:09.55 | nick125_lappy | i also wonder, with a asterlink line, is it possible to have multiple in and out lines (i know asterlink supports 6, but, is there something special i have to do to asterisk?) |
| 05:09.56 | jsmith | BrainSurg: IAX and SIP termination... |
| 05:10.00 | JerJer | jsmith: trying to finish members.nufone.net, but its not getting there very quickly |
| 05:10.01 | adelas | heh lol |
| 05:10.01 | hellop | my asterisk is behind my firewall, I want to use neighbor's wireles with my VOIP wifi phone. Aparently that's not possible. |
| 05:10.11 | h3x | this is so funny |
| 05:10.18 | h3x | http://www.sylantro.com/partners.html |
| 05:10.27 | h3x | I like how they have like the whole entire fortune 500 listed there |
| 05:10.44 | h3x | and then on the right side theres a place setter "Customer quotes or facts will go here" |
| 05:10.52 | h3x | hahahhahahahhahahha |
| 05:11.11 | JerJer | that's great |
| 05:11.16 | BrainSurg | :) |
| 05:11.23 | JerJer | this page needs to be mirrored for future reference |
| 05:11.26 | hellop | So maybe I need to get an IAX wifi phone instead of a sip one? Has anyone played with wireless VOIP and trying to roam? |
| 05:11.35 | h3x | and then they flipped horizontal |
| 05:11.42 | nick125_lappy | anyone here use asterlink? |
| 05:11.43 | hellop | BTW, I do have permission from 3 different wireless access points to experiment with. |
| 05:11.44 | h3x | that stupid clip art |
| 05:12.26 | h3x | hellop: bwahaha theres no iax wireless phones |
| 05:13.00 | h3x | makarios.blessed.net - - [18/Aug/2005:22:07:32 -0700] "CONNECT 66.103.132.86:11111 HTTP/1.0" 501 16 "-" "unknown" |
| 05:13.00 | h3x | makarios.blessed.net - - [18/Aug/2005:22:07:32 -0700] "POST /default.rxml HTTP/1.0" 200 4874 "-" "unknown" |
| 05:13.03 | h3x | what a jackass |
| 05:13.03 | JerJer | correction, there are no available iax wireless phones |
| 05:13.17 | h3x | well |
| 05:13.18 | hellop | h3x HAHA! probably not even possible to run on a PDA.. |
| 05:13.27 | h3x | i guess you could hook up a cordless phone to an iaxy and you have one |
| 05:14.02 | BrainSurg | What's the going rate for an IAXy? |
| 05:14.10 | h3x | $TOO_MUCH |
| 05:14.10 | JerJer | oh there is one - they are just being very lets say cautious with the release |
| 05:14.10 | jsmith | BrainSurg: Around $100. |
| 05:14.27 | h3x | for $80 you can get a two line device from sipura that actually has good audio quality and lots of features. |
| 05:14.35 | h3x | oh and t.38 is in beta now |
| 05:14.36 | jsmith | h3x: But it's SIP. |
| 05:14.43 | JerJer | h3x: obviously you haven't seen the new iaxy |
| 05:14.45 | h3x | so? it does 10 flavors of nat traversal |
| 05:14.57 | h3x | its just a new case!@ |
| 05:15.00 | JerJer | um no |
| 05:15.02 | jsmith | h3x: I'd still take a new IAXy... |
| 05:15.11 | jsmith | h3x: Speak what you know... |
| 05:15.24 | h3x | the only point of iax is nat traversal |
| 05:15.30 | h3x | but the sipura works anyway |
| 05:15.35 | h3x | i triple natted it and it still worked |
| 05:16.05 | jsmith | h3x: Wrong again! |
| 05:16.11 | JerJer | iax uses less bandwidth :P |
| 05:16.18 | h3x | no it dosent |
| 05:16.19 | jsmith | h3x: There's much more to IAX than just NAT traversal. |
| 05:16.23 | h3x | you cant run g.729 on the iaxy ! |
| 05:16.23 | JerJer | yes it does |
| 05:16.27 | JerJer | there is no RTP |
| 05:16.29 | jsmith | h3x: Try trunking SIP calls... |
| 05:16.37 | JerJer | hence less overhead |
| 05:16.39 | h3x | jsmith: You cant use trunking on a single line device |
| 05:16.41 | JerJer | hence less bandwidth |
| 05:16.51 | h3x | another problem |
| 05:16.55 | jsmith | h3x: I realize that... I *INVENTED* IAX trunking... |
| 05:17.00 | h3x | theres no way to keep someone from iaxyprov'ing your iaxy |
| 05:17.06 | flewid | ManxPower: thanks for your suggestion earlier today of adding txgain in the zap cfg to stop the double digits :) working great |
| 05:17.13 | flewid | manxpower: owe ya a beer :) |
| 05:17.21 | BrainSurg | What is T.38? |
| 05:17.23 | h3x | my point is you gain nothing from getting a iax capible ATA |
| 05:17.33 | JerJer | its iax |
| 05:17.36 | JerJer | not sip |
| 05:17.39 | jsmith | h3x: And again, I'll say that I'd much prefer one that does IAX, and not SIP |
| 05:17.39 | h3x | so |
| 05:17.47 | JerJer | jsmith: ditto |
| 05:17.49 | h3x | iax will never do t.38 |
| 05:17.49 | jsmith | h3x: For some of us, it's a big deal. |
| 05:17.57 | JerJer | iax doesn't need T.38 |
| 05:18.06 | jsmith | h3x: What does T.38 have to do with the transport protocol :-) |
| 05:18.18 | jsmith | Fax over IP is so lame anyway... |
| 05:18.20 | hellop | who was the actor in the move Quicksilver? |
| 05:18.28 | hellop | movie |
| 05:18.28 | JerJer | if you want to fax, download coppice's kick ass dsp code and fax |
| 05:18.29 | jsmith | ... hello, twenty-first century here ... |
| 05:18.40 | JerJer | then transfer it over IP using a binary transfer protocol |
| 05:18.45 | SplasPood | wtf is the point of fax over IP anyway |
| 05:18.47 | JerJer | then shit it out a PRI at the far end |
| 05:18.55 | JerJer | instant T.37 without the bullshit |
| 05:19.03 | h3x | yeah well think about what ATAs are used for, not everybody wants to throw their fax machine away |
| 05:19.05 | jsmith | SplasPood: Amen! |
| 05:19.22 | JerJer | nobody says they have to throw away their fax machine |
| 05:19.22 | SplasPood | jsmith: no come on.. there's gotta be an application.... |
| 05:19.23 | jsmith | h3x: Have you ever actually tried sending a lot of faxes over T.38? |
| 05:19.45 | jsmith | h3x: I'd rather have appendicitis. |
| 05:19.48 | JerJer | store and forward is so much more efficient |
| 05:20.02 | h3x | yeah but then youd need like 8 megs of ram in your ata |
| 05:20.08 | jsmith | SplasPood: Stopgap technology? |
| 05:20.12 | JerJer | who says the ata has to do shit? |
| 05:20.28 | h3x | lets see |
| 05:20.28 | SplasPood | jsmith: I suppose... |
| 05:20.34 | h3x | most people use ATAs on cable modems |
| 05:20.36 | JerJer | ata - local asterisk box running rxfax |
| 05:20.47 | h3x | since for DSL you need a analog line in 90% of the world |
| 05:20.49 | h3x | or more |
| 05:20.51 | jsmith | Later all... have a good night, and dream of starfish... |
| 05:20.54 | ilustrate | what happens if the power goes out with asterisk. does it reset after you restart the computer. |
| 05:20.58 | h3x | and nothing else is practical in most situations |
| 05:21.01 | JerJer | local asterisk box running rxfax scp's tiff image to remote aasterisk box with txfax and pri |
| 05:21.09 | h3x | cable modems have packet loss out the yinyang |
| 05:21.12 | JerJer | remote asterisk box with txfax and pri shits it out the PRI |
| 05:21.12 | jsmith | ilustrate: Only if you tell it to... it's just a program, like any other... |
| 05:21.36 | ilustrate | so it should just start where it left off? |
| 05:21.40 | h3x | yeah right, sell that idea to somebody doing vonage type shit |
| 05:21.48 | JerJer | don't do vonage type shit |
| 05:21.52 | jsmith | ilustrate: Does your word processor start where it left off? |
| 05:22.03 | JerJer | be smarter than Citron |
| 05:22.05 | JerJer | come on |
| 05:22.05 | jsmith | ilustrate: It will only restart on reboot if you set it up that way... |
| 05:22.17 | ilustrate | no. but i guess i meant. would i have to re-configure everything? |
| 05:22.22 | hellop | We just want to be able to bust out the PDA, which can get net access from any free access point, and work with the * box specified in the softphone. Any suggestions? Anyone want to work with me on this? |
| 05:22.28 | jsmith | ilustrate: No. |
| 05:22.29 | JerJer | ilustrate: this is why we have hard drives |
| 05:22.35 | ilustrate | o.k. |
| 05:22.47 | jsmith | G'night all... |
| 05:22.49 | BrainSurg | Anyways, good night everyone. |
| 05:22.50 | JerJer | night |
| 05:22.51 | hellop | I got PDAs, CF wireless cards, Wireless fones, ATA's whooo wants to help? |
| 05:22.54 | *** part/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 05:23.08 | Hogie | okay hellop, bring them all to my house |
| 05:23.10 | Hogie | and I'll help |
| 05:23.14 | hellop | I get free access points at work, at home and at the college. |
| 05:23.31 | h3x | there is a ton of shit the sipuras do |
| 05:23.53 | hellop | Hogie, how about you fly to Hawaii and U can stay at my house by the beach? |
| 05:23.54 | h3x | like, codecs besides g.721 and g.726?!?? |
| 05:24.10 | *** join/#asterisk tc0nn (n=tec@spyglass.timsnet.com) |
| 05:24.15 | JerJer | codec selection will change |
| 05:24.18 | Hogie | hellop: na |
| 05:24.26 | h3x | with a two bit half ass processor? |
| 05:24.33 | JerJer | and locking down the provisioning will too |
| 05:24.57 | JerJer | sipura's run on a DVD based MPEG microcontroller |
| 05:25.01 | Hogie | JerJer: you mean since linksys bought sipura? |
| 05:25.01 | JerJer | that's not two bit ? |
| 05:25.13 | JerJer | Hogie: i'm talking about the iaxy |
| 05:25.15 | tc0nn | Linksys..ie Cisco? |
| 05:25.31 | h3x | they still have the best designed and best priced atas there is |
| 05:25.37 | JerJer | in your opnion |
| 05:25.59 | Hogie | best ata = asterisk box with a fxs card;) |
| 05:26.17 | h3x | well lets see |
| 05:26.24 | tc0nn | Any idea what mutes DTMF when connected to the console/dsp (via oss channel) |
| 05:26.35 | JerJer | the mute function? |
| 05:26.51 | tc0nn | Is it the oss driver killing dtmf? |
| 05:27.06 | h3x | $200 for two ports of iaxy |
| 05:27.11 | h3x | or $85 for two ports of sipura |
| 05:27.11 | tc0nn | Yeah, I get a quick burp of DTMF, not enough to detect though. |
| 05:27.13 | h3x | DUH |
| 05:27.21 | JerJer | sipura doesn't do IAX |
| 05:27.27 | JerJer | at least publicly |
| 05:27.38 | h3x | but theres no reason to do iax |
| 05:27.39 | h3x | ? |
| 05:27.40 | SplasPood | Unlimited NationalAccess/BroadbandAccess services cannot be used (1) for uploading, downloading or streaming of movies, music or games, (2) with server devices or with host computer applications, including, but not limited to, Web camera posts or broadcasts, automatic data feeds, Voice over IP (VoIP), automated machine-to-machine connections, or peer-to-peer (P2P) file sharing, or (3) as a substitute or backup for private lines or dedicated data c |
| 05:27.47 | JerJer | sure there is |
| 05:27.48 | SplasPood | hahaha... thats verizon's wireless broadband TOS |
| 05:27.52 | h3x | Like what |
| 05:28.06 | JerJer | because its not sip |
| 05:28.09 | JerJer | or H.323 |
| 05:28.15 | JerJer | or mcgp |
| 05:28.20 | h3x | so what |
| 05:28.20 | tc0nn | or skinny |
| 05:28.23 | Hogie | SplasPood: you wouldn't want to do any of those anyway |
| 05:28.26 | h3x | it still uses more bandwidth when you have a crappy ass codec to use |
| 05:28.37 | Hogie | the latency is so freaking high on BroadbandAccess it sucks |
| 05:28.39 | JerJer | that will change, so your argument is moot |
| 05:28.42 | Hogie | even ssh is laggy |
| 05:28.51 | h3x | without a hardware change? |
| 05:28.54 | tc0nn | not as band as starband |
| 05:29.00 | SplasPood | Hogie: I heard it was still better than anything else around... |
| 05:29.12 | JerJer | i don't work for Digium, so i duno |
| 05:29.14 | h3x | if digium charged the $10 per channel license for g.729 |
| 05:29.19 | Hogie | it is... i have a pc card |
| 05:29.20 | h3x | that means it would be $110 for a iaxy with g.729 |
| 05:29.22 | Hogie | with it |
| 05:29.36 | *** join/#asterisk B4 (n=B4@202.69.48.245) |
| 05:29.42 | JerJer | the $10 is for the codec implementation inside of asterisk |
| 05:29.49 | JerJer | and that's retail price |
| 05:29.54 | JerJer | who pays retail ? |
| 05:29.57 | Hogie | but there's no way teamspeak will even work on it, I dont know how you would live with 300 - 400ms latency and IAX2/SIP protocol on it |
| 05:30.00 | B4 | hi |
| 05:30.38 | Hogie | I even tried uploading pics from webcam on the laptop before while driving down the road, and even at 30 secs, it was having problems getting overloaded at times |
| 05:30.50 | B4 | can anyone suggest what would be the best way to hookup a PRI (Euro) to asterisk? |
| 05:30.53 | *** join/#asterisk sangee (n=rkuru@Toronto-HSE-ppp3697175.sympatico.ca) |
| 05:31.00 | Hogie | with an E1? |
| 05:31.00 | JerJer | yeah GPRS is not friendly to anything cool, yet |
| 05:31.02 | tc0nn | B4- a digium card |
| 05:31.08 | tc0nn | TE410P |
| 05:31.14 | JerJer | TE411P |
| 05:31.23 | JerJer | its worth it |
| 05:31.35 | tc0nn | I'd stay away from the 411 for now |
| 05:31.36 | B4 | what about the new dual span cards? |
| 05:31.49 | JerJer | tc0nn: why you say that? we use them in production |
| 05:31.52 | *** part/#asterisk ilustrate (i=user@200.92.29.182) |
| 05:31.56 | tc0nn | I've got two... with the echo cards removed. Box is unstable with them on. |
| 05:32.04 | sangee | anyone know how to setup realtime sip registration using mysql? someone please help me? |
| 05:32.06 | JerJer | rock solid here |
| 05:32.16 | tc0nn | What hw/os ? |
| 05:32.43 | JerJer | dell with e1000 disabled |
| 05:32.49 | JerJer | linux 2.4.20 |
| 05:33.01 | Hogie | 1850? |
| 05:33.07 | B4 | any experience with TE210P? |
| 05:33.15 | JerJer | sc 1425 |
| 05:33.19 | JerJer | i think |
| 05:33.25 | JerJer | something like that |
| 05:33.26 | tc0nn | Well.. I've got one Dell 1750 w/RHES4, another 1750 with RHES3, neither have the e1000 disabled. Its on a different IRQ anyway. |
| 05:33.44 | JerJer | get RES shit off of there |
| 05:33.47 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:33.53 | JerJer | go with slack or debian - good lord |
| 05:34.10 | Hogie | gentoo 4evah! |
| 05:34.26 | JerJer | tc0nn: doesn't matter...for some reason Digium's cards hate the e1000 |
| 05:34.30 | B4 | JerJer: any experience with TE210P? |
| 05:34.33 | JerJer | no matter what IRQ |
| 05:34.37 | JerJer | no we have all 4 span cards |
| 05:35.24 | tc0nn | Well, its been up and running ever since we removed the echo card. I may try a pci nic and put the card back in. |
| 05:35.32 | JerJer | or the e1000 hates digium - haven't figured it out exactly |
| 05:35.48 | tc0nn | digium wanted me to do that so they could get a developer in the box to see wtf was going on. |
| 05:35.52 | B4 | I am just 1 km from telco exchange, can run the PRI straight on copper without HDSL right? |
| 05:36.00 | tc0nn | no |
| 05:36.02 | JerJer | tc0nn: yep |
| 05:36.10 | JerJer | i know the drill |
| 05:36.14 | Goshen | Where does monitor dump the recordings? |
| 05:36.28 | B4 | no? |
| 05:36.34 | JerJer | <PROTECTED> |
| 05:36.36 | JerJer | i'm guessing |
| 05:36.37 | tc0nn | B4. I think t1 only goes 384 feet on cat5, probably less on a telco bundle. |
| 05:36.38 | h3x | the nice thing about having sip and rtp seperated is if you get your DIDs from 3rd parties, you can set up the RTP session directly between a consumer on the internet and that underlying provider without chewing up your own bandwidth |
| 05:36.46 | h3x | and keep an accurate CDR of the time logged |
| 05:36.51 | JerJer | don't buy from 3rd parties |
| 05:36.58 | B4 | E1? |
| 05:37.19 | tc0nn | e1 probably less... |
| 05:37.25 | tc0nn | Since its 25% more bw. |
| 05:37.38 | h3x | all for just managing call setup traffic |
| 05:37.45 | B4 | hmm ok, so would need the HDSL on both ends |
| 05:37.49 | B4 | more cost! |
| 05:37.57 | tc0nn | Why is that? |
| 05:38.14 | tc0nn | Should be able to find old/unused smartjacks in any office building... |
| 05:38.25 | B4 | well, I would need to buy the equipment |
| 05:38.31 | JerJer | h3x: so then how do you garuntee QoS that way? |
| 05:38.41 | B4 | smartjacks? |
| 05:38.41 | *** join/#asterisk dev2005 (n=dev2003@222.33.36.205) |
| 05:38.42 | h3x | how do you guarantee QoS when your customer isnt on your network |
| 05:38.45 | h3x | it dsoent fucking matter |
| 05:38.51 | JerJer | sure it does |
| 05:38.57 | JerJer | its called peering |
| 05:39.13 | tc0nn | smartjacks/hygain unit/xDSL to T1 converting testpoint/demarc.. whatever you want to call it. |
| 05:39.18 | JerJer | when you get peering agreements you make damn sure they pass on QoS information |
| 05:39.26 | h3x | well the places i get dids on voip are on better internet connections than i could ever afford |
| 05:39.39 | JerJer | you hope |
| 05:39.48 | Vco | heh..heh.. |
| 05:39.50 | tc0nn | h3x - who are you using? |
| 05:39.53 | tc0nn | for did's? |
| 05:39.54 | h3x | :P |
| 05:39.56 | JerJer | i don't base my business on hope |
| 05:40.09 | h3x | a half dozen different clecs |
| 05:40.28 | syle | Set option TDD MODE, value: OFF(0) on Zap/3-1 |
| 05:40.30 | h3x | www.carrierone.net/dids has a excel spreadsheet |
| 05:40.32 | tc0nn | How much do they charge? Just curious if I'm getting ripped off. |
| 05:40.34 | syle | whats this mean |
| 05:40.36 | h3x | although a lot of stuff is missing from it |
| 05:41.08 | tc0nn | What kind of circuit was on that zap channel? |
| 05:41.09 | h3x | they are all different ,and its a apples to oranges comparison because what im buying them for wholesale requires a $5k-$25k a month committment |
| 05:41.16 | JerJer | tc0nn: if you are paying per minute, you are getting ripped off |
| 05:41.20 | syle | ILEC line |
| 05:41.23 | h3x | shit |
| 05:41.24 | h3x | no |
| 05:41.30 | h3x | you get ripped off if its flat rate |
| 05:41.38 | tc0nn | no, I pay $10 per hundred per month. |
| 05:41.44 | h3x | because what happens is they turn off simultaenous call capability |
| 05:41.49 | JerJer | um |
| 05:41.53 | JerJer | you buy UNE |
| 05:41.57 | *** join/#asterisk oej (n=oej@ti111210a080-0697.bb.online.no) |
| 05:42.06 | h3x | oh you are talking about DIDs on a PRI or what? |
| 05:42.10 | tc0nn | yes |
| 05:42.15 | tc0nn | DID on PRO |
| 05:42.16 | tc0nn | PRI |
| 05:42.17 | h3x | no im talking about vlip delivery |
| 05:42.17 | JerJer | trunks are separate yo |
| 05:42.22 | tc0nn | ok |
| 05:42.23 | h3x | voip |
| 05:42.41 | JerJer | that's the VoIP provider versus real telco carrier play right there |
| 05:42.43 | Goshen | what file has the moh default music? |
| 05:42.52 | h3x | for instance i pay KMC $0.002/minute or they want $2/mo for a flat rate did with 500 min cap |
| 05:43.03 | JerJer | which is highway robbery |
| 05:43.11 | h3x | anybody with a fucking calculator can figure out |
| 05:43.13 | tc0nn | <PROTECTED> |
| 05:43.14 | h3x | 500 minutes is $1 |
| 05:43.14 | dev2005 | who use E400P ? |
| 05:43.27 | Goshen | tcc0nn: what tarball has the default moh mp3 files? |
| 05:43.41 | JerJer | we pay $45 a month for a OC-3 cross connect |
| 05:43.42 | h3x | it isnt that bad with little hidden fees |
| 05:43.52 | tc0nn | cvs asterisk-sounds probably |
| 05:44.05 | Goshen | I just installed asterisk on my new system and it doesn't have any files in /var/spool/asterisk/moh |
| 05:44.08 | Goshen | I did asterisk-sounds |
| 05:44.19 | tc0nn | upload your favorite mp3 then |
| 05:44.23 | tc0nn | and make mpg123 |
| 05:44.32 | syle | you mean /var/lib/asterisk/sounds lol |
| 05:44.40 | h3x | considering their whole market coverage is tier 3 markets |
| 05:44.41 | JerJer | syle: not for MOH |
| 05:44.43 | tc0nn | not for moh |
| 05:44.46 | h3x | that price is nothing |
| 05:44.56 | h3x | thats places that level 3 wants like 0.03/Minute |
| 05:44.58 | h3x | well |
| 05:45.01 | h3x | 2.something anyway |
| 05:45.01 | Goshen | my last install had those nice moh files |
| 05:45.14 | JerJer | h3x: we don't pay per minute for our DIDs |
| 05:45.17 | JerJer | period |
| 05:45.22 | h3x | and they are using telica switches |
| 05:45.42 | JerJer | and just because we are only offering DIDs in one state doesn't mean we don't have more states |
| 05:45.44 | h3x | yeah well they dont do usage sensitive when you are doing TDM for everything |
| 05:45.58 | tc0nn | Whats GlobalCrossing like these days? They've been cold-calling the shit out of me.. |
| 05:46.02 | JerJer | its the right way to do it |
| 05:46.12 | JerJer | reselling someone elses VoIP is just asking for trouble |
| 05:46.16 | h3x | it dosent matter how you do it when your carrier uses VoIP anyway and double converts |
| 05:46.28 | JerJer | don't buy from carriers that do that shit |
| 05:46.38 | h3x | all of them are using voip now. |
| 05:46.43 | JerJer | um no |
| 05:47.24 | h3x | The larger ones using sonus gear has already replaced IMT links on the back end of their legacy switches with voip |
| 05:47.30 | h3x | to conserve bandwidth between sites |
| 05:47.42 | JerJer | who says we have to use larger ones? |
| 05:47.43 | tc0nn | Anyone know if a sip.conf -> realtime upload script exists? |
| 05:47.48 | h3x | so its already going through a g.729 compression cycle once |
| 05:48.21 | h3x | Because all the smaller ones are just dumping their traffic on qwest, mci, global crossing, sprint, at&t anyway? |
| 05:48.31 | JerJer | um i am talking about DID yo |
| 05:48.41 | h3x | Uh.... |
| 05:48.55 | h3x | DIDs transported for long distances need echo cans |
| 05:49.02 | h3x | on TDM |
| 05:49.08 | JerJer | ok and the problem is ? |
| 05:49.21 | h3x | well everybodys depreciating that shit including global crossing, focal, etc. |
| 05:49.23 | h3x | even mci |
| 05:49.38 | JerJer | and who says they need to be transported long distances using TDM? |
| 05:49.44 | h3x | the MRCs are rediculious |
| 05:49.51 | h3x | well then you have to buy colo everywhere otherwise |
| 05:50.02 | JerJer | ok and the problem is? |
| 05:50.22 | BharatS | Hello there |
| 05:50.28 | JerJer | moo |
| 05:50.30 | h3x | the point is that their million dollar sonus, telica, whatever gear is a hell of a lot better quality than your shitty asterisk boxes at doing echo cancelling and codec translation |
| 05:50.45 | BharatS | I am working on building a Manager interface for the Asterisk |
| 05:50.46 | JerJer | if you say so |
| 05:50.53 | tc0nn | Cool. We need one of those. |
| 05:51.09 | JerJer | yet what kind of technical complaints can you find about us? |
| 05:51.09 | h3x | for example, on a properly configured asterisk box with a t1 card in it connected to a PRI to a CLEC here |
| 05:51.24 | h3x | with echo cans disabled as they should be, g.711 fax over a LAN to that box i cant get more than 9600 bps |
| 05:51.37 | JerJer | asterisk was not designed with data in mind |
| 05:51.44 | h3x | but g.711 over the public internet, many hops away to their east coast gateways even i get 14.4kbps no problem |
| 05:51.51 | h3x | it aint data, its sound |
| 05:51.51 | BharatS | can anybody suggest me the some key features that are need to be considered in building the ASterisk Manager interface |
| 05:52.00 | h3x | if my fax machine can notice a sound problem across a lan |
| 05:52.08 | h3x | then its not doing a very good job with voip translation to tdm |
| 05:52.36 | h3x | if it has to renegoate to 9600 bps |
| 05:52.51 | JerJer | i'm done feeding the troll |
| 05:52.53 | JerJer | goodnight |
| 05:52.56 | h3x | hahahah |
| 05:53.05 | h3x | well asterisk isnt exactly quality software |
| 05:53.13 | JerJer | then don't fucking use it |
| 05:53.24 | h3x | nor is linux either, but lots of religious nuts to go around |
| 05:53.41 | JerJer | yet you benifit from both on a per minute bases |
| 05:53.48 | h3x | im using SER |
| 05:53.54 | JerJer | who isn't? |
| 05:54.10 | bonez41 | what's SER, at the risk of sounding ignorant? |
| 05:54.17 | h3x | well i have only 1 asterisk box for transporting calls, and thats to handle two local PRIs thats it |
| 05:54.27 | tc0nn | Sip proxy |
| 05:54.31 | h3x | everything else is offloaded directly to a carrier's voip gateway or to MAX TNTs |
| 05:54.47 | fad | openH323 and oh323 are same? i want to install openh323 with asterisk |
| 05:54.50 | h3x | so im not really using it that much :P |
| 05:55.09 | JerJer | you are still benefiting from it |
| 05:55.16 | h3x | fad save yourself some trouble by using neither |
| 05:55.17 | bonez41 | h3x, why not writing your own, something better? |
| 05:55.18 | JerJer | yet you bitch at it every change you get |
| 05:55.18 | tc0nn | single point of failure |
| 05:55.19 | h3x | I am? |
| 05:55.46 | fad | i have tp installed |
| 05:55.46 | h3x | I was just discussing the use of it in a "ITSP call transport" type application |
| 05:55.58 | tc0nn | toilet paper? |
| 05:56.00 | JerJer | calls go in and/or our of your two local PRIs don't they? |
| 05:56.01 | h3x | or using it instead of big irons |
| 05:56.12 | Goshen | found the sample moh files /asterisk-1.0.9/sounds/fpm-calm-river.mp3 |
| 05:56.14 | *** join/#asterisk santiago (n=santiago@63.245.86.202) |
| 05:56.16 | Goshen | not sure why they didn't install |
| 05:56.21 | h3x | very few, and i am going to move those over to tnt's t1 ports when i get more dsps in a box |
| 05:56.40 | JerJer | until then, stop bitching |
| 05:56.40 | syle | is it stupid to set echo cancellation on a fxs port hehe |
| 05:56.46 | h3x | when my xspedius contract expires im porting everythign to KMC |
| 05:56.56 | tc0nn | I'm out.. Have fun guys |
| 05:56.56 | h3x | because xspedius cant do E911 MSAG records on a per DID basis |
| 05:57.04 | *** part/#asterisk tc0nn (n=tec@spyglass.timsnet.com) |
| 05:57.08 | h3x | then it will be completely irrelevant |
| 05:58.09 | h3x | I'm not really bitching about anything, other than pointing out that I don't see how taking TDM into a ESP's network and converting to VoIP with $0 software just so you can "maintain QoS" |
| 05:58.21 | h3x | really achieves anything over using all the carrier's expensive gateways |
| 05:58.30 | JerJer | lets see- what bout application logic |
| 05:59.09 | BharatS | has anybody worked on the Asterisk Manager Interface? |
| 05:59.10 | JerJer | or i want to see a carrier gateway become a ham radio repeater controller |
| 05:59.15 | BharatS | please reply |
| 05:59.20 | h3x | when * really costs more per channel than second hand equipment with DSPs and G.729 licenses built in |
| 05:59.22 | JerJer | or control X-10 devices |
| 05:59.35 | JerJer | or interface the telephone with the web |
| 05:59.44 | Insanity5 | Has anyone uses didx.org? |
| 05:59.57 | JerJer | Insanity5: no - you shouldn't either |
| 05:59.59 | BharatS | nop |
| 06:00.00 | h3x | well of course its the best thing around for end user applications but that isnt what nufone nor carrier one does |
| 06:00.12 | h3x | i mean, objectively |
| 06:00.16 | Insanity5 | JerJer - It seems the biggest porblem would be provider reliability and latency. |
| 06:00.16 | Insanity5 | hehe. |
| 06:00.17 | BharatS | setting up gui for the Asterisk... |
| 06:00.43 | JerJer | h3x: i've configured thousands of end-user asterisk boxes |
| 06:00.56 | JerJer | and have written hundereds of custom apps for end-user asterisk boxes |
| 06:00.59 | JerJer | how is that not what we do? |
| 06:01.05 | Insanity5 | Those are some pretty bad sample on hold music files. |
| 06:01.11 | h3x | shrug i guess its the chicken and the egg |
| 06:01.30 | Insanity5 | Anyone have a good greeting/on hold .wav/.mp3 colelction? :P |
| 06:01.40 | Insanity5 | I thought about ripping one of the tracks off the half life 2 soundtrack... hehe |
| 06:01.52 | JerJer | Insanity5: have The Voice record your own custom prompts |
| 06:02.05 | Insanity5 | the voice? |
| 06:02.10 | JerJer | Alison |
| 06:02.16 | Hogie | we got mr moviephone to do our prompts |
| 06:02.17 | h3x | i don't believe voip belongs on the public internet either, but if its gonna end up there then it dosent really matter where it comes from that much |
| 06:02.22 | h3x | its all finger pointing from there |
| 06:02.23 | Insanity5 | JerJer - whois alison? |
| 06:02.30 | JerJer | The Voice |
| 06:02.32 | h3x | (back to my point about offloading RTP) |
| 06:02.36 | Insanity5 | huh? |
| 06:02.53 | Hogie | http://www.digium.com/index.php?menu=product_detail&category=extras&product=THEVOICE |
| 06:02.57 | Hogie | there Insanity5 |
| 06:03.03 | JerJer | the voice! |
| 06:03.13 | h3x | obviously if someone called and bitched that they cant reach some gateway |
| 06:03.32 | h3x | all youd have to do is change an option to proxy that call over |
| 06:03.48 | JerJer | not if your DIDs are coming from that one gateway |
| 06:03.57 | h3x | so in a sense you sorta have a means of redundancy workarounds |
| 06:04.02 | Insanity5 | JerJer- No sample files :( |
| 06:04.18 | JerJer | she records custom prompts |
| 06:04.26 | h3x | well, most of these providers have both direct trunking into here and public ip access |
| 06:04.35 | Hogie | Insanity5: you know the demo file? |
| 06:04.41 | h3x | if i change where i tell them to send the call to it will come in on a private ip circuit |
| 06:04.42 | Hogie | er, the demo dialplan? |
| 06:04.45 | JerJer | operative word being 'most' |
| 06:05.01 | Insanity5 | Hogie - which one? |
| 06:05.10 | Insanity5 | JerJer - I thoguht some sexy lady would be the best bet. hehe |
| 06:05.14 | h3x | most being all but the morons like Level(3) which serves areas that nobody else does |
| 06:05.18 | Hogie | Insanity5: when you do make samples after compiling * |
| 06:05.20 | h3x | so if a subscriber had a number with them |
| 06:05.23 | h3x | through anybody |
| 06:05.27 | h3x | they would have the exact same problem |
| 06:05.44 | Insanity5 | Hogie - Are tehy .wav.mp3 files somewhere? |
| 06:05.48 | JerJer | not if you pulled it off of the tadem |
| 06:05.54 | h3x | er. i forgot to mention they dont have a private IP MPLS whatever option |
| 06:06.02 | Hogie | gsm, in /var/lib/asterisk/sounds |
| 06:06.12 | Insanity5 | Hogie - Gas, winamp can't play those, can it? |
| 06:06.21 | Hogie | there's a gsm plugin for it |
| 06:06.22 | Hogie | so yes |
| 06:06.25 | h3x | "pulling it off the tandem" in the 9000 rate centers i can cover right now with 6 carriers is seemingly impossible |
| 06:06.34 | JerJer | if you say so |
| 06:06.57 | *** join/#asterisk Blake0PS (n=blake@blakeops.com) |
| 06:07.00 | h3x | that would be like at least 6,000 boxes that cost $1500-2000k a piece or something |
| 06:07.07 | h3x | er $2000 |
| 06:07.07 | JerJer | no |
| 06:07.47 | h3x | pulling off the tandem huh, are you using ss7 box or what |
| 06:07.58 | Blake0PS | Can Asterisk detect ACTS tones (from Bell System payphones)? |
| 06:08.17 | h3x | oh my god |
| 06:08.27 | h3x | i hope you are setting up COCOTs |
| 06:08.37 | Blake0PS | death to COCOTs |
| 06:08.41 | Insanity5 | Hogie - God, the quality of those gsm files sucks. |
| 06:08.49 | h3x | are you writing an article for 2600 or something? |
| 06:09.00 | Insanity5 | Do pay phones even exist anymore? |
| 06:09.10 | h3x | somebody needs to make a voip payphone |
| 06:09.14 | Blake0PS | I use payphones instead of a cell phone, it's cheaper. |
| 06:09.15 | JerJer | they already have |
| 06:09.18 | h3x | or at least an ATA that does ground start |
| 06:09.22 | Insanity5 | Blake0PS - IF there's any left. |
| 06:09.49 | h3x | all the payphone guys have gotten into ATMs instead |
| 06:09.56 | h3x | i went to this stupid payphone show here in vegas |
| 06:10.05 | h3x | all of them are has beens |
| 06:10.14 | h3x | talking about the old days of payphones |
| 06:10.15 | Insanity5 | I really don't know if "the voice" is that great. |
| 06:10.23 | Insanity5 | Maybe it'd sound better if it wasn't crappy gsm. |
| 06:10.32 | Blake0PS | I'm a phone phreak, there are payphones here, and I want to know if Asterisk can reliably detect ACTS tones |
| 06:10.39 | h3x | Insanity5: theres a web site with a whole bunch of voice recording talent out there |
| 06:10.41 | nick125_lappy | time to redo my Customer Care/Support menu...this should be fun |
| 06:10.45 | Goshen | I saw a payphone the other day, I had to look twice because it was an unusual sight |
| 06:10.46 | h3x | allison is on it among others |
| 06:10.53 | Insanity5 | h3x - Somebodyh as to have a repository of common items :) |
| 06:11.00 | Insanity5 | hehe |
| 06:11.00 | h3x | if you just google for allison smith you will run across it |
| 06:11.11 | JerJer | or google john todd |
| 06:11.25 | JerJer | he has a sounds repository of alison prompts |
| 06:11.33 | Blake0PS | Here are the payphones in my area http://www.yapl.org/list.php?list=user&show=cataloged&uid=9&perpage=1000 |
| 06:11.35 | *** join/#asterisk ryansc (n=ryansc@c-67-164-188-180.hsd1.co.comcast.net) |
| 06:11.36 | h3x | blake__: So, why would anybody have ever written the code to do that |
| 06:11.39 | h3x | er |
| 06:11.40 | Insanity5 | Is it better quality if you don't use gsm? |
| 06:11.49 | JerJer | a lot of which made it into asterisk-addons, but there are some that didn't |
| 06:11.55 | JerJer | er asterisk-sounds |
| 06:11.58 | JerJer | whatever its caleld |
| 06:11.59 | JerJer | called |
| 06:12.11 | h3x | you know whats funny, i accidently made a $13,500 aculab board a redbox once |
| 06:12.30 | h3x | speaking of asterisk-addons |
| 06:12.39 | h3x | the MYSQL() dialplan thing sucks ass |
| 06:12.42 | Blake0PS | ACTS tone detection is useful. These phone exist and Asterisk is a PBX. |
| 06:12.47 | h3x | i should write a new one |
| 06:12.59 | h3x | but maybe PGSQL() |
| 06:13.09 | Juggie | how about just use agi |
| 06:13.12 | h3x | blake, how would you plug a payphone into asterisk |
| 06:13.34 | Blake0PS | FXS? |
| 06:13.44 | h3x | Juggie, well thats just it, i want to avoid doing that when its goign to get hammered with requests to fill in dialplan variables |
| 06:13.58 | Insanity5 | Do you need a license to enter the payphone business or can you put your own outside your gas station? |
| 06:14.01 | Juggie | you can set a dialplan var direct from within agi |
| 06:14.04 | h3x | blake, its ground start |
| 06:14.26 | Insanity5 | Althoguh, I doubt you could net even $5/month :P |
| 06:14.29 | Insanity5 | hehe |
| 06:14.38 | h3x | I know you can but calling an AGI, or using MYSQL(), forks a threaded process from hell, ergo Asterisk |
| 06:14.47 | h3x | to exec an external program |
| 06:14.47 | Blake0PS | oh, no FXS modules do ground start? |
| 06:14.53 | h3x | and it cant keep the mysql socket open |
| 06:14.55 | h3x | to reuse |
| 06:15.06 | h3x | well on channel banks yes |
| 06:15.13 | Juggie | h3x, theres a PHP() as well |
| 06:15.14 | h3x | but ive never seen a ATA that does ground start |
| 06:15.15 | *** join/#asterisk Twister (i=Twister@68-117-151-227.dhcp.chtn.wv.charter.com) |
| 06:15.17 | Insanity5 | Would these voice prompts sound better if they weren't gsm? |
| 06:15.42 | h3x | juggie, I just wonder what happens when fork gets called 200 times at the same time |
| 06:15.54 | h3x | that would suck <TM> |
| 06:16.03 | Juggie | h3x, i believe that would be called bad design on your part. |
| 06:16.22 | Juggie | you cant have it both ways |
| 06:16.26 | Juggie | fork per MYSQL |
| 06:16.28 | Juggie | or blocking |
| 06:16.29 | Juggie | pick one |
| 06:16.38 | Juggie | if you share a socket, then MYSQL() calls are blocking |
| 06:16.41 | h3x | neither. how about app_mysql_that_works_for_shit.so |
| 06:16.46 | Insanity5 | Here's a bunch of free voice prompts: http://www.microsoft.com/downloads/details.aspx?FamilyID=ef62d47d-ce5a-44fa-864f-3c31c14769b7&displaylang=en |
| 06:16.51 | h3x | I might actually call it that |
| 06:16.51 | Insanity5 | =) |
| 06:16.58 | h3x | when the module is loaded, it creates a pool of mysql handles |
| 06:17.10 | Juggie | h3x, its going to fork, or be blocking. |
| 06:17.16 | h3x | no it wont |
| 06:17.21 | h3x | mysql library is thread safe |
| 06:17.49 | h3x | do the whole thing in C as an asterisk module |
| 06:18.20 | Juggie | btw, how would you generate 200 calls a second |
| 06:18.28 | h3x | it aint that hard |
| 06:18.30 | Juggie | er, 200 calls to the fork |
| 06:18.38 | Juggie | well, how many lines do you have |
| 06:18.44 | h3x | hahaha |
| 06:18.45 | Juggie | or whats your max active calls |
| 06:18.49 | h3x | see www.carrierone.net |
| 06:18.59 | h3x | actually it wont be used here but |
| 06:19.16 | Juggie | ok |
| 06:19.20 | Juggie | so give me a number |
| 06:19.24 | Juggie | max calls on the system |
| 06:19.24 | h3x | asterisk can handle a couple thousand calls on one box if its all voip |
| 06:19.33 | Juggie | hahaha |
| 06:19.34 | Juggie | you wish |
| 06:19.37 | h3x | it does |
| 06:19.41 | h3x | no codec translation |
| 06:19.45 | h3x | no zap channels |
| 06:19.53 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
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| 06:19.59 | Juggie | as long as * isnt in the RTP path |
| 06:20.01 | Juggie | you will be ok |
| 06:20.07 | Juggie | maybe |
| 06:20.14 | h3x | that dosent really matter if its copying |
| 06:20.22 | h3x | but yeah better without |
| 06:20.23 | h3x | in any case |
| 06:20.26 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
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| 06:20.33 | Juggie | linux isnt the best kernel for packet throughput |
| 06:20.34 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
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| 06:20.39 | h3x | the point is, to make it work for everybodys applications better not just mine |
| 06:20.41 | Juggie | you would do better without |
| 06:20.41 | Insanity5 | What format should on hold music be in? |
| 06:21.15 | h3x | needing to manipulate data in mysql or postgres or whatever is important in some situations to do for every single call through the system |
| 06:21.19 | Juggie | h3x, regardless, you could create a MYSQL with connection pooling yes. |
| 06:21.24 | h3x | sometimes several times for a single call |
| 06:21.30 | Goshen | After leaving a voicemail which is supposed to get mailed (using old config files on a new system) I get this error message |
| 06:21.30 | Goshen | postdrop: warning: unable to look up public/pickup: No such file or directory |
| 06:21.32 | *** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
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| 06:21.41 | h3x | surely i hope that cdr mysql and so forth do it this way |
| 06:21.43 | Goshen | is that a postfix configuration problem? |
| 06:21.45 | Juggie | h3x, then i must ask you, why dont you do all your work from within one script |
| 06:21.59 | h3x | because, it may be at different call stages |
| 06:22.04 | Juggie | your point? |
| 06:22.07 | Juggie | put it all in one agi. |
| 06:22.32 | Juggie | agi can contain an entire call flow from answer to hangup |
| 06:22.33 | h3x | because the dialplan cant move along |
| 06:22.36 | h3x | if you are in an agi |
| 06:22.45 | Juggie | you dont need dialplan within an agi |
| 06:22.48 | Juggie | you can code a dialplan |
| 06:22.53 | Juggie | in perl, or php or whatever |
| 06:23.05 | nick125_lappy | anyone here know a cheap ATA that is unlocked and works nicely with asterisk (under $50 maybe??) |
| 06:23.14 | h3x | i guess but some of the stuff needed may not be supported in the agi |
| 06:23.16 | Insanity5 | What's a good cheap voip phone or ATA? It seems like the adapter is the better way to go with more flexibility, but on hold and stuff might be an issue. |
| 06:23.26 | Juggie | h3x, give me an example. |
| 06:23.34 | h3x | or the dialplan needs to be manipulated using sql storage |
| 06:23.40 | h3x | by outside programs |
| 06:23.53 | Juggie | what do you mean |
| 06:24.03 | h3x | Well ill just make up something |
| 06:24.10 | h3x | lets say you were to set up a vonage wannabe co |
| 06:24.22 | h3x | (bad example from what iw as just talking about but whatever) |
| 06:24.43 | h3x | you have an extranet that creates new shit in your dialplan via sql depending on what your users select for options |
| 06:25.06 | h3x | meanwhile in your dialplan it may need to do sql queries to grab settings such as, |
| 06:25.07 | Juggie | uhhuh |
| 06:25.13 | h3x | if some feature is enabled or not |
| 06:25.16 | Juggie | let me show you an example |
| 06:25.20 | h3x | or, what your forward to phone number is |
| 06:25.27 | h3x | or whatever. ... |
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| 06:25.54 | h3x | it would just make so much more sense if you could use MYSQL() to grab that and stuff it in a variable at will |
| 06:26.03 | h3x | if it would scale like that |
| 06:26.32 | Juggie | take a look @ |
| 06:26.53 | h3x | I realize you can do this with an AGI, but the thing is that with something that is so pivotal in any given dynamically configurable application |
| 06:26.54 | Juggie | www.pastebin.ca/20399 |
| 06:27.02 | h3x | its time well spent to make fucking sql queries inside the dialplan work |
| 06:27.07 | Juggie | they do work |
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| 06:27.55 | h3x | i have my doubts about asterisk forking a couple dozen times in a second and not screwing up something |
| 06:28.20 | Juggie | so then reduce the forking |
| 06:28.32 | Juggie | fork once for an agi |
| 06:28.35 | Juggie | or fix mysql |
| 06:28.37 | h3x | you cant if thats because of a couple dozen calls set up at once |
| 06:28.59 | h3x | the other problem is that mysql will shit itself |
| 06:29.03 | h3x | if you connect to it too many times |
| 06:29.49 | h3x | it sure would be nice to rewrite asterisk |
| 06:30.04 | h3x | in an interpeted language with C modules doing the tough shit on the back end |
| 06:30.13 | crash3m | talking about resperl? |
| 06:30.14 | Juggie | holy crap dude |
| 06:30.17 | Juggie | mysql will be fine |
| 06:30.25 | Juggie | php connects to mysql every web hit |
| 06:30.39 | h3x | i thought php used a persistant connection pool |
| 06:30.40 | Juggie | and servers get hundreds, of hits a second... it will be ok. |
| 06:30.48 | Juggie | not unless you explicitally enable it |
| 06:30.51 | h3x | oh |
| 06:30.59 | h3x | hmmm |
| 06:31.00 | crash3m | I've seen resperl self-fuck an * box |
| 06:31.00 | *** join/#asterisk MustangMatt (n=Miranda@firewall.thoughtprocess.net) |
| 06:31.09 | h3x | heh |
| 06:31.17 | Juggie | i did a hammer/bulkdial a few weeks ago |
| 06:31.18 | Juggie | using * |
| 06:31.22 | Juggie | with mysql cdr enabled |
| 06:31.26 | h3x | ahahaha |
| 06:31.32 | Juggie | i crashed the pbx providing me the t1 |
| 06:31.34 | Juggie | * was ok |
| 06:31.37 | crash3m | heh |
| 06:31.43 | Juggie | i actually locked the line card on the other end |
| 06:31.45 | h3x | that is amazing |
| 06:31.49 | Juggie | it needed a hard reset |
| 06:31.58 | h3x | when i started using asterisk a couple years ago |
| 06:32.08 | h3x | i simply hooked it up back to back to an aculab card |
| 06:32.09 | h3x | PRI |
| 06:32.15 | h3x | told the aculab card to place 23 calls at once |
| 06:32.19 | h3x | and the asterisk box.. well.... |
| 06:32.23 | h3x | the fucking kernel paniced |
| 06:32.38 | h3x | things have come a ways |
| 06:32.47 | h3x | but still |
| 06:32.58 | h3x | it could be better. |
| 06:33.11 | h3x | HEAD crashes a lot |
| 06:33.18 | syle | SetVar(loop=loop+1) |
| 06:33.20 | syle | how do i do this |
| 06:33.24 | h3x | and that just comes with any huge ass C program |
| 06:33.28 | Qwell | syle: do what? |
| 06:33.34 | syle | increment var by 1 |
| 06:33.55 | Qwell | SetVar(loop=$[${loop} + 1]) |
| 06:33.56 | Qwell | maybe? |
| 06:34.15 | h3x | but if the crazy shit went into glue modules and the logic was all written in interpeted languages |
| 06:34.40 | h3x | I did this exact same thing with my aculab software development, i wrapped its libraries on the C backend of an interpeted language Pike |
| 06:35.01 | h3x | after i got all those bugs worked out i never had any crash problems anymore |
| 06:35.06 | Juggie | h3x, gdb the crashes, post patches.... people will love you |
| 06:35.18 | h3x | because Pike's garbage collection took care of a lot of things |
| 06:35.42 | h3x | but it could be done on anything.. Lua, Ruby, Perl, Python, Whatever |
| 06:36.04 | h3x | even better is have it hinged on one thing and then glue a bunch of interpeters together so you can code in anything |
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| 06:37.20 | h3x | it sounds really sick but i think it would work |
| 06:38.19 | Juggie | agi allows any intrepreter |
| 06:38.30 | h3x | i know :P |
| 06:38.58 | *** join/#asterisk MmmmToop (n=michael@rndf-146-26-134.telkomadsl.co.za) |
| 06:39.34 | h3x | hehe ruby would be a crazy language to write a pbx in |
| 06:39.54 | Juggie | fyi... * has a new lang for dialplan |
| 06:39.55 | Juggie | ael |
| 06:40.06 | h3x | adv extension logic? |
| 06:40.13 | Juggie | maybe |
| 06:40.14 | Juggie | not sure |
| 06:40.37 | h3x | The point of making most of the pbx in an interpeted language is you can inherit objects and modify them |
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| 06:41.03 | h3x | from a 3rd party development standpoint anyway |
| 06:41.16 | *** part/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
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| 06:41.56 | h3x | it would be a pain in the ass to decide what language to use though |
| 06:42.18 | h3x | python is organized and popular now but its slow as hell |
| 06:42.32 | h3x | which normally isnt a big deal but is for this sort of thing |
| 06:43.26 | MustangMatt | After all the help drray gave me last night getting asterisk setup and working I decided to writeup an article on how to setup asterisk from scratch using debian. I tried to write it from a beginner's point of view. Would anyone be interested in reading what I've got and giving me feedback? |
| 06:44.24 | h3x | i hate to tell you this but i think that already exists :) |
| 06:44.27 | MustangMatt | More specifically, it's how to setup asterisk as an answering machine. |
| 06:44.37 | h3x | oh |
| 06:44.38 | MustangMatt | h3x: I was unable to find one as simple as mine. |
| 06:46.01 | h3x | well i would but i dont use debian so.. |
| 06:46.03 | MustangMatt | Anyway here's what I've got. http://kaatman.com/asteriskansweringmachine.html I'd love feedback. It's still missing a giant section on extensions.conf and doesn't achieve all the goals I setup yet. Think of it as written for newbs by a newb. |
| 06:51.24 | Guggemand | hmm, my * installation doesnt seem to retry a sip register when it fails |
| 06:53.29 | Blake0PS | what part of a call to an asterisk box detects and decodes DTMF to a digital digit? |
| 06:54.15 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 06:57.33 | X-Rob | Blake0PS - None of it. |
| 06:58.11 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 07:03.43 | Blake0PS | notice i asked what part of a call, not what part of asterisk |
| 07:04.01 | Blake0PS | something has to convert my 2 frequencies making up the digit 9 to an analog form |
| 07:07.51 | X-Rob | Blake0PS - perhaps more information. Describe, exactly, where you're pushing this nine. |
| 07:08.57 | Blake0PS | I want to find out where the A to D conversion occurs so that I can detect my own analog signal and convert it to something digital |
| 07:09.14 | Blake0PS | ACTS tones to be specific |
| 07:09.32 | Guggemand | if you have your phone connected to some box i would think that converts it |
| 07:09.50 | Guggemand | if its a softphone it sends the digital when you push it |
| 07:09.52 | X-Rob | *picks one of the many places it could be, because you're not being anythign near 'exactly'* In the phone |
| 07:10.31 | Guggemand | if its a phone not connected to anything, then its magic :) |
| 07:11.27 | Blake0PS | ACTS phone -> CO -> FXO -> Asterisk |
| 07:12.00 | X-Rob | Ahha |
| 07:12.06 | X-Rob | so it's coming in via a X100 or TDM? |
| 07:12.22 | Blake0PS | something of that flavour |
| 07:12.45 | X-Rob | Sorry |
| 07:13.03 | X-Rob | 'How is it coming in. Via a X100 or a TDM?' |
| 07:13.19 | X-Rob | the answer is 'X100' 'TDM' or 'Something else' and you then explain what the somethign else is |
| 07:13.22 | X-Rob | why are you being so damn stupid? |
| 07:13.45 | Blake0PS | ask a stupid question... |
| 07:13.56 | Blake0PS | "Which are you using, X100P or TDM" |
| 07:14.13 | Guggemand | Blake0PS if you want help you should answer in a sane way |
| 07:14.18 | Blake0PS | The answer is TDM with FXO module |
| 07:14.41 | X-Rob | OK. TDM's do DTMF detection in hardware. |
| 07:15.14 | X-Rob | however, if you do care about significantly clever DSP stuff, look at spandsp, which does soft faxing via asterisk |
| 07:15.35 | Blake0PS | what FXO interfaces have software DTMF detection? |
| 07:16.06 | X-Rob | X100 |
| 07:16.11 | X-Rob | (as far as I know) |
| 07:16.29 | X-Rob | some channel banks have hardware, some channel banks don't |
| 07:16.31 | X-Rob | look at libpri |
| 07:17.35 | X-Rob | no |
| 07:17.38 | X-Rob | don't look at libpri |
| 07:18.00 | X-Rob | look at dsp.c |
| 07:18.07 | X-Rob | (in /usr/src/asterisk) |
| 07:18.07 | X-Rob | sorry |
| 07:18.41 | Blake0PS | aha |
| 07:18.48 | Blake0PS | thanks |
| 07:20.28 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@ip68-109-146-168.ph.ph.cox.net) |
| 07:21.14 | *** join/#asterisk BeaveIPAQ (n=BeaveIPA@adsl-214-72-186.jax.bellsouth.net) |
| 07:22.53 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-14-177.cybersurf.com) |
| 07:28.22 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 07:28.23 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 07:31.23 | *** join/#asterisk oej (n=oej@194.248.94.59) |
| 07:38.57 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:40.12 | *** join/#asterisk jackthe (n=jesse@d594f03e.ftth.concepts.nl) |
| 07:41.28 | djin_ib | Does anyone know a reason for 'HDLC Abort (6)' after upgrading from 1.0.7 to 1.0.9(.1)? |
| 07:41.47 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-147-53.sw.res.rr.com) |
| 07:41.59 | djin_ib | ManxPower suggested avoiding onboard NIC yesterday. |
| 07:42.38 | djin_ib | However, this is not easy for me to try (remote location) |
| 07:43.27 | *** join/#asterisk Sloboda (n=sloboda@195.137.227.70) |
| 07:46.10 | nighty- | uhmm |
| 07:48.00 | Sloboda | Hi! Could you tell me your preferences for linux soft phone for *. I have tried khone and linphone. Need only dialing buttons and phone book. |
| 07:48.52 | Sloboda | Is one with more pretty interface? |
| 07:49.28 | Sloboda | Like ExpressTalk, for example. (It is only for Win) |
| 07:53.11 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 07:53.44 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 07:54.51 | *** part/#asterisk HellAgony (i=HellAgon@200.121.236.192) |
| 07:58.40 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 08:01.23 | *** join/#asterisk nagl (n=nagl@137.208.4.172) |
| 08:02.37 | *** join/#asterisk ukh (n=ukh@130.226.34.10) |
| 08:03.12 | *** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121) |
| 08:03.35 | ukh | is there any particular IRC channel where people hang out discussing non-Asterisk SIP/H.323/VoIP issues with servers, proxies, gateways and clients? |
| 08:04.43 | Sloboda | REGISTER 123 |
| 08:05.10 | crash3m | ukh: /msg chanserv list *voip* |
| 08:05.54 | *** part/#asterisk Sloboda (n=sloboda@195.137.227.70) |
| 08:06.22 | crash3m | is it just me, or is voip-info.org going really slow right now? |
| 08:07.54 | crash3m | wait, it might just be the nightly crons starting |
| 08:10.23 | gordonjcp | all over the world, it's time for someone's nightly cron to start |
| 08:12.16 | nighty- | gordonjcp: not here |
| 08:12.44 | *** join/#asterisk edwin_ (i=edwin@252-131-222-203.rev.techex.net.au) |
| 08:13.00 | gordonjcp | nighty-: nor here |
| 08:13.24 | *** join/#asterisk wd (n=wd@pdc.investtel.cz) |
| 08:13.25 | gordonjcp | it's pretty slow here too |
| 08:14.29 | nighty- | gordonjcp: it's more because it is day time |
| 08:14.37 | nighty- | gordonjcp: like 10:00 AM |
| 08:14.38 | nighty- | :) |
| 08:14.45 | gordonjcp | 9am here |
| 08:15.01 | gordonjcp | it may be 3am where the servers are, prime time for cron.daily |
| 08:15.07 | nighty- | gordonjcp: where ? |
| 08:15.17 | nighty- | gordonjcp: UK ? |
| 08:15.42 | gordonjcp | I'm in the UK, I don't know where the servers are |
| 08:15.45 | gordonjcp | presumably the US |
| 08:16.19 | gordonjcp | -5 would put them in eastern US, at about 3am |
| 08:16.48 | *** join/#asterisk Liquefact (n=NNSCRIPT@62.116.83.61) |
| 08:17.05 | nighty- | gordonjcp: LA |
| 08:17.31 | gordonjcp | ok |
| 08:17.40 | gordonjcp | western US, 1am or so? |
| 08:17.48 | nighty- | gordonjcp: PNAP , LAX |
| 08:17.58 | nighty- | gordonjcp: they are BBN customers |
| 08:18.14 | nighty- | gordonjcp: for IP transit |
| 08:18.52 | *** join/#asterisk gaffer (n=cliff@69.36.245.165) |
| 08:18.56 | nighty- | gordonjcp: commpartners I mean |
| 08:19.07 | gordonjcp | ah right |
| 08:19.37 | Liquefact | is there anyone there who can answer my questions about asterisk |
| 08:19.55 | *** part/#asterisk gaffer (n=cliff@69.36.245.165) |
| 08:20.20 | gordonjcp | Liquefact: well, since you haven't told us what your questions are, there's no way of knowing |
| 08:20.38 | Liquefact | ok, i'll tell |
| 08:21.18 | *** join/#asterisk dudes (n=dudes@12-215-34-84.client.mchsi.com) |
| 08:21.30 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 08:21.38 | Liquefact | i'm to set up an intercom system which will work in a simulator |
| 08:21.38 | Lan16`spdy^gTp | hello |
| 08:21.50 | Lan16`spdy^gTp | i've a problem can somebody help me ? |
| 08:21.56 | Liquefact | can asterisk be used in this system as an intercom |
| 08:22.15 | Lan16`spdy^gTp | odbc say that its connected to my mysql database |
| 08:22.16 | gordonjcp | Liquefact: depends what you want it to do |
| 08:22.33 | Lan16`spdy^gTp | extconfig.conf is set up |
| 08:22.34 | dudes | Liquefact - I believe you can use console/default as an intercom ... (via your soundcard.) |
| 08:22.46 | Liquefact | two way voice communication between consoles, and nothing more |
| 08:22.54 | Lan16`spdy^gTp | and for iax ive no registration for peer |
| 08:22.56 | Lan16`spdy^gTp | :s |
| 08:23.05 | Lan16`spdy^gTp | somebody now the problem ? |
| 08:23.40 | Liquefact | but the communication must be digital, recordable, and traceble |
| 08:24.03 | dudes | Liquefact - that's what the wiki is for |
| 08:24.06 | dudes | use it |
| 08:24.06 | *** join/#asterisk Inv_arp (i=junya@adsl-8-230-188.mia.bellsouth.net) |
| 08:24.25 | Lan16`spdy^gTp | dudes can you help me with the odbc support |
| 08:24.31 | Lan16`spdy^gTp | everything is good |
| 08:24.34 | Lan16`spdy^gTp | odbc loaded |
| 08:24.38 | Lan16`spdy^gTp | connected to database |
| 08:24.45 | Lan16`spdy^gTp | extconfig is set up |
| 08:24.56 | dudes | Lan16`spdy^gTp - I read what you typed before |
| 08:25.04 | Lan16`spdy^gTp | ok |
| 08:25.09 | Lan16`spdy^gTp | and you think ? |
| 08:25.17 | Liquefact | in asterisak home page this channel is considered as a support channel |
| 08:25.23 | dudes | I don't use odbc |
| 08:25.35 | Lan16`spdy^gTp | ive try to use mysql only |
| 08:25.41 | Lan16`spdy^gTp | in result with the cvs |
| 08:25.54 | Lan16`spdy^gTp | the phone don't ring etc... |
| 08:25.55 | Liquefact | that's why i'm asking questions here, you may not reply me and so not bothered |
| 08:26.08 | dudes | Liquefact - read the fucking wiki |
| 08:26.15 | dudes | you'll find what you need |
| 08:26.16 | Lan16`spdy^gTp | authentification with mysql is good but everything go down when i make a call |
| 08:26.31 | Lan16`spdy^gTp | its wy i dont want to use the cvs |
| 08:26.50 | Lan16`spdy^gTp | i use it 3 times and 3 times it give me problem |
| 08:26.51 | Lan16`spdy^gTp | :s |
| 08:26.53 | dudes | Liquefact - it's your fault you're too lazy to read |
| 08:27.01 | tzafrir_laptop | it is a free support channel: we help if we feel like it. Mostly if the question is interesting enough |
| 08:27.14 | Liquefact | fuck you dudes |
| 08:27.20 | dudes | tzafrir_laptop - hehe |
| 08:27.30 | Lan16`spdy^gTp | ^^ |
| 08:27.34 | dudes | Liquefact - you're not my type ... |
| 08:27.49 | Lan16`spdy^gTp | can someone have got some idea with odbc support |
| 08:28.09 | syle | is there a digit to insert a pause is dialing DTMF tones? |
| 08:28.32 | syle | in |
| 08:28.45 | Liquefact | everywhere in the net there are bastards like you trying to yell to people trying to learn something |
| 08:28.46 | drumkilla | dudes: if you don't want to help, don't answer the question. |
| 08:29.20 | *** join/#asterisk meppl (n=mephisto@84-245-169-70.ipool.celox.de) |
| 08:29.58 | Lan16`spdy^gTp | Liquefact whats your problem ? |
| 08:30.03 | dudes | drumkilla - I told him how he could achieve his goal; all he has to do is read like the rest of us. |
| 08:30.48 | syle | how do you insert a pause in a dial sequence? anyone know? |
| 08:30.58 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:30.59 | Lan16`spdy^gTp | no |
| 08:31.28 | dudes | digital ... (asterisk is a software PBX?) ... traceable (of course) ... recordable (yes) |
| 08:31.54 | whisker | style: use 'w' characters; ie Dial(Zap/1/1234www111) |
| 08:33.25 | Lan16`spdy^gTp | why odbc dont request my database however odbc is connected to :^o ? |
| 08:34.04 | *** part/#asterisk ukh (n=ukh@130.226.34.10) |
| 08:34.27 | tzafrir_laptop | Liquefact, what do you mean by "traceable"? |
| 08:35.43 | dudes | Liquefact - I will not respond to your last statement (but fuck you) ... you can use console/default as a intercom. You can use several ways to trace via asterisk. you can use monitor to record any use of the intercom extension. Look I helped. |
| 08:36.10 | tzafrir_laptop | Liquefact, Asterisk uses by default a rather low-quality "phone-line" encoding for sound |
| 08:36.34 | tzafrir_laptop | Is that quality good enough? |
| 08:37.13 | dudes | You can use System to echo to a file via use of an extension. Asterisk also provides other methods (cdr) and so on |
| 08:37.30 | tzafrir_laptop | I still don't see how you can trace (regardless of Asterisk), unless you have some other identification method there |
| 08:38.07 | dudes | you can set a var via a channel (callerid) and append it to a file |
| 08:38.26 | tzafrir_laptop | dudes, but this is just an intercom, right? |
| 08:38.38 | dudes | he wants it traceable |
| 08:38.39 | tzafrir_laptop | what generates the caller id in the first place? |
| 08:38.41 | dudes | lets not forget |
| 08:38.47 | dudes | the calling in party |
| 08:39.01 | dudes | everything coming into a asterisk PBX is identified |
| 08:39.02 | tzafrir_laptop | Isn't it an intercom device? |
| 08:39.15 | dudes | man you're high |
| 08:39.27 | dudes | a channel is bridged |
| 08:39.32 | dudes | to the intercom extension |
| 08:39.42 | dudes | therefore, you can trace it if you want to |
| 08:40.18 | dudes | a intercom can be connected to your soundcard via a PA system and have an extension set to it. But you can trace it (if you want to) |
| 08:41.05 | RoyK | cypromis: ping |
| 08:41.06 | Liquefact | thanks all, i think i got the answer |
| 08:41.12 | dudes | Good |
| 08:41.27 | Liquefact | sorry for all bad words, and taking time |
| 08:41.35 | dudes | do you want an example of how you can make it traceable via a working example |
| 08:41.43 | dudes | I'd be glad to PM you |
| 08:41.56 | dudes | It's not like I don't do this 16-18hr's a day for a living |
| 08:42.16 | Liquefact | i have little irc knpwledge, and dont know what PM is |
| 08:42.24 | Liquefact | also my english is not that good |
| 08:42.26 | dudes | Private Msg |
| 08:42.34 | Liquefact | yep, sure |
| 08:42.43 | tzafrir_laptop | ~pm |
| 08:42.43 | jbot | extra, extra, read all about it, pm is private message, or perl mongers, or pathetic moron: when you see someone say pm, they're asking if you think that they're a pathetic moron, or something you don't do without asking permission |
| 08:43.20 | dudes | jbot knows all |
| 08:43.20 | jbot | and don't you forget it |
| 08:43.21 | tzafrir_laptop | Liquefact, you can also /msg jbot pm |
| 08:43.39 | tzafrir_laptop | this will avoid the whole channel hearing about it |
| 08:44.17 | tzafrir_laptop | ~all |
| 08:44.18 | *** join/#asterisk juraj (i=juraj@smtp.rebelium.net) |
| 08:44.31 | juraj | hello, anyone has any idea what this means? |
| 08:44.34 | juraj | Aug 18 21:20:09 WARNING[2179] channel.c: No channel type registered for 'zap' |
| 08:44.34 | juraj | Aug 18 21:20:09 WARNING[2179] app_meetme.c: Unable to open pseudo channel - trying device |
| 08:44.50 | juraj | is it only that it uses ztdummy device? |
| 08:45.06 | Lan16`spdy^gTp | why odbc dont request my database however odbc is connected to :^o ? |
| 08:45.08 | tzafrir_laptop | juraj, chan_zap not loaded? meetme needs a timing source? |
| 08:45.09 | *** join/#asterisk r0m (n=SysOp@bl8-28-55.dsl.telepac.pt) |
| 08:45.15 | r0m | a |
| 08:45.18 | juraj | tzafrir_laptop: I have ztdummy |
| 08:45.28 | juraj | tzafrir_laptop: it works actually :) |
| 08:45.35 | juraj | tzafrir_laptop: I'm just curious about that error |
| 08:45.36 | tzafrir_laptop | juraj, maybe chan_zap.so is not loaded? |
| 08:45.37 | r0m | good morning |
| 08:45.40 | juraj | or warning |
| 08:45.41 | tzafrir_laptop | zap show channels |
| 08:45.47 | juraj | tzafrir_laptop: I don't want chan_zap |
| 08:46.00 | juraj | tzafrir_laptop: I'm a sip only pbx |
| 08:46.07 | juraj | tzafrir_laptop: :) |
| 08:46.13 | tzafrir_laptop | you need it to use ztdummy for timing |
| 08:46.16 | *** join/#asterisk esi (n=ewaldirc@simonis.xs4all.nl) |
| 08:46.34 | juraj | tzafrir_laptop: I actually use ztdummy and have no chan_zap, since the meetme works |
| 08:46.39 | *** join/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com) |
| 08:46.47 | juraj | tzafrir_laptop: I have no problem with the conference, just curious about the warnings |
| 08:47.28 | Lan16`spdy^gTp | why odbc dont request my database however odbc is connected to :^o, someone can help me ? |
| 08:47.39 | dudes | juraj - if you use conf you use chan_zap and the psuedo zap device |
| 08:48.55 | juraj | *CLI> zap show channels |
| 08:48.55 | juraj | No such command 'zap' (type 'help' for help) |
| 08:48.59 | juraj | :) |
| 08:49.26 | juraj | ok, so I presume it says, it tried the zap pseudochannel and if it could not load it (since chan_zap is not loaded) it tried the ztdummy pseudo device? |
| 08:50.01 | tzafrir_laptop | dudes, or maybe meetme reads from psedudo directly and not through chan_zap? |
| 08:50.21 | juraj | tzafrir_laptop: it has to, because I really have no chan_zap and it works :) |
| 08:50.40 | dudes | usarules1*CLI> zap show channels |
| 08:50.41 | dudes | <PROTECTED> |
| 08:50.41 | dudes | <PROTECTED> |
| 08:50.48 | juraj | I'm just curious, I'm trying to debug a bug that has happened to me yesterday |
| 08:50.51 | juraj | http://bugs.digium.com/view.php?id=4986 |
| 08:50.52 | juraj | :) |
| 08:51.29 | juraj | dudes: it seems, that if chan_zap is not loaded, it uses the device directly |
| 08:51.38 | juraj | dudes: and that's what the warning is about |
| 08:51.45 | *** join/#asterisk MustangMatt (n=Miranda@firewall.thoughtprocess.net) |
| 08:51.53 | dudes | then use chan_zap an resovle it |
| 08:52.08 | juraj | no problem, so this is not a problem at all |
| 08:52.19 | juraj | the device works |
| 08:52.20 | dudes | Since the ztdummy is a *dummy* drive for zap |
| 08:52.46 | *** join/#asterisk Kernel_Core (i=Raph@217.218.94.242) |
| 08:53.03 | Kernel_Core | hi all , anybody familiar to configure X101P card ?! |
| 08:53.21 | juraj | what happened to me yesterday, that two people had a conference in a room and after cca 15 minutes, one got disconnected by asterisk |
| 08:53.26 | juraj | I'm trying to figure out why |
| 08:53.52 | juraj | one more question about the notice :) |
| 08:53.53 | juraj | Aug 18 21:17:01 NOTICE[2179] chan_sip.c: stale nonce received from '<sip:1003@192.168.1.222>;tag=as3c32c77f' |
| 08:53.55 | *** join/#asterisk nagl (n=nagl@137.208.4.162) |
| 08:53.57 | dudes | juraj - I have 10 lines on a PBX and conferences last sometimes for hours a day ... no problems |
| 08:53.57 | juraj | what does this mean? |
| 08:54.12 | juraj | dudes: this is quite a complicated setup |
| 08:54.16 | juraj | dudes: :) |
| 08:54.17 | dudes | haha |
| 08:54.24 | dudes | you think you have a complicated setup |
| 08:54.26 | dudes | I laugh |
| 08:54.32 | dudes | haha |
| 08:54.35 | juraj | dudes: :) |
| 08:54.45 | juraj | dudes: ok, this one is not that complicated on a pbx side |
| 08:55.10 | juraj | dudes: it goes through a checkpoint firewall in one IPSec connection and then it uses another ipsec encapsulated in it to the PBX |
| 08:55.18 | juraj | dudes: :-) |
| 08:55.38 | Kernel_Core | is zaptel 1.0.9.1 compatible with X101P devices ? Like V.92 Intel softmodems ? |
| 08:55.38 | dudes | it sounds like you make things too complicated for it's own good |
| 08:55.46 | juraj | dudes: it took me quite a lot of time (including fixing bugs in racoon) to make this work. |
| 08:55.55 | juraj | dudes: it's not what I want :) |
| 08:56.14 | juraj | dudes: it's what a client wants. they absolutely require (by secure policy) to use checkpoint to enter the LAN |
| 08:56.20 | juraj | dudes: no other way to get there |
| 08:56.35 | juraj | dudes: and they need all clients protected by ipsec using their own certificates |
| 08:56.44 | juraj | dudes: nothing I can change about it |
| 08:56.46 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 08:56.56 | juraj | dudes: but it seems I found at least one bug in asterisk thanks to this :) |
| 08:57.05 | Mimmus | what's the exact meaning of 'overlapdial=yes' |
| 08:57.13 | dudes | perhaps it's a bug on your end |
| 08:57.27 | juraj | i'm sure there's also a bug on my end |
| 08:57.44 | juraj | but this one: http://bugs.digium.com/view.php?id=4986 is certainly in asterisk :) |
| 08:58.16 | dudes | it's not like asterisk is written is C or anything (not that I have anything against C.) |
| 08:58.24 | dudes | Or it's a program that is free |
| 08:59.25 | *** join/#asterisk razu (n=razu@217-159-242-106-dsl.est.estpak.ee) |
| 08:59.46 | juraj | it's free and it's in c, i don't get what you tried to tell :) |
| 09:00.02 | *** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin) |
| 09:00.05 | PakiPenguin | hello everyone |
| 09:00.10 | juraj | hello PakiPenguin |
| 09:00.11 | juraj | :) |
| 09:00.24 | dudes | juraj - sorry ... I'm pretty pissed because I spend the hole day today fixing shit for retards. (and I had a date, but I had to work and fix shit.) |
| 09:00.37 | PakiPenguin | dudes, i feel for you :( |
| 09:01.48 | dudes | juraj - the fact is it's C (C is a good language for somethings.) and as good as C is it has it's flaws. And, moreover, it's free so dont' expect the world. |
| 09:02.09 | juraj | dudes: np. I'm trying to finish this stupid project for two weeks and I always fix bugs. Two days ago, it was some stupid bug in racoon and Linux kernel, that has been well-known for at least half a year. Up until now, the kernel developers and the racoon developers are telling, that the bug is on the other's side, so no one fixed it. I prototyped a working fix in twenty minutes. If they at least said that on a page or something.... |
| 09:02.10 | dudes | that's my point |
| 09:02.47 | juraj | ah |
| 09:02.47 | juraj | :) |
| 09:03.15 | juraj | dudes: I'm quite familiar with the source code to some extend, at least with chan_sip.c. But for some internals I am not. |
| 09:03.24 | dudes | juraj - as a dev my self ... sometimes things don't get done because other things on a list are more important. But not always =)( |
| 09:03.58 | dudes | juraj - so you have an issue with conference's getting disconnected |
| 09:04.06 | juraj | yep |
| 09:04.09 | dudes | after XX amount of time |
| 09:04.26 | dudes | because your clients are using IPSec |
| 09:04.33 | juraj | yep. the network connection stays alive. I can ping the asterisk machine. |
| 09:04.40 | juraj | I don't know if it's because of that. |
| 09:05.29 | dudes | is there sometype of timeout on the IPsec DB |
| 09:06.09 | juraj | the issue I reported in Mantis is, that if the client disconnects, chan_sip sends the data, because in ipsec setup, you always get temporary error if there's no security association. But this one is not related, I just reported it because in the morning, I found 400MB log of "Resource temporarily unavailables" :) |
| 09:06.17 | dudes | I suppose I should look over your bug report |
| 09:06.39 | juraj | dudes: it is indeed. several timeouts. There's an SA timeout and a rekeying timeout. |
| 09:07.00 | juraj | dudes: anyways, when new SA is registered, the old one can be used too |
| 09:07.09 | dudes | I've never seen a asterisk log file over 131MB's |
| 09:07.10 | dudes | haha |
| 09:07.15 | juraj | dudes: so you don't get even one packet lost |
| 09:07.42 | dudes | and that was on a Dual P4 3.06Ghz /w HT |
| 09:07.44 | juraj | each SA has expiration time and before it ends, new SA is negotiated and both are valid. |
| 09:08.32 | dudes | so is this bug report your's? |
| 09:08.55 | juraj | dudes: yeah, the bug report is that when you get "Resource temporarily unavailable"s for whole night, it's not _that_ temporary, and I wonder if someone would be interested in RTP packet if the whole night lasting temporary error would get resolved :) |
| 09:09.03 | juraj | yep, this one is mine :) |
| 09:09.18 | juraj | but that does not solve my problem, even if it's resolved :)))) |
| 09:09.30 | dudes | are you allowing the RTP ports from IPSec to IPSec |
| 09:10.09 | dudes | ie. 10000-20000 to communicate between the two? (bare in mind I'm hammered.) |
| 09:10.51 | juraj | dudes: allowing everything |
| 09:11.18 | juraj | dudes: (if they use ipsec, but that's not based on port numbers, it's a security policy set using setkey to the kernel:) |
| 09:11.20 | dudes | Not to be rude ... but I heard that line before with IPsec techs and it was horseshit |
| 09:11.23 | juraj | Aug 18 22:18:16 WARNING[887] app_meetme.c: Unable to write frame to channel: Success |
| 09:11.42 | juraj | what does this mean? :) |
| 09:11.51 | *** join/#asterisk Rowters (n=SilverDr@dsl-201-129-88-148.prod-infinitum.com.mx) |
| 09:12.18 | juraj | dudes: yeah, maybe, but now it works for me |
| 09:12.32 | dudes | I spend a day with a client resovling an issue similiar to yours ... anyway it was because they're techs were idiots |
| 09:12.51 | juraj | dudes: but at least this one is open-source |
| 09:12.56 | dudes | Not saying you are |
| 09:13.03 | juraj | dudes: checkpoint is similiar piece of shit, but I can't even find/fix bugs in the source |
| 09:13.06 | hellop | Anyone ever change the idle picture on a Polycom IP phone? |
| 09:13.07 | juraj | :))) |
| 09:13.30 | hellop | I tried to do it using a gimp created "Windows BMP" maybe I have to use an acutal windows one? |
| 09:14.12 | *** join/#asterisk secure75 (n=mic@p549A3EF7.dip0.t-ipconnect.de) |
| 09:14.35 | juraj | dudes: I'm pretty sure the ipsec is quite okay now. I don't get a one lost ping, nothing. |
| 09:14.46 | dudes | if a packet timesout |
| 09:14.58 | dudes | it's clearly dropping somewhere |
| 09:15.06 | juraj | dudes: and even if I did, I believe several lost packets must not be problem for a voip connection |
| 09:16.07 | juraj | aaah, I hate this stupid ipsec setup :) |
| 09:16.13 | dudes | no shit |
| 09:16.15 | dudes | haha |
| 09:16.22 | dudes | just use iptables |
| 09:16.51 | juraj | for what? :) |
| 09:17.15 | juraj | you mean firewall instead of encryption and signing? :) |
| 09:18.23 | dudes | argue with me about firewalls and encrytion when I'm sober and can type |
| 09:18.51 | dudes | regardless ... why encrytpe conf data |
| 09:18.53 | kyoo | I'm trying to understand digium hardware... I can today buy a TDM01B and later upgrade it to a TDM03B or a TDM22B myself by buying modules? |
| 09:19.50 | tzafrir_laptop | hellop, why use the gimp for conversions when use can use imagemagick's convert? |
| 09:19.56 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 09:20.45 | dudes | because gimp is better |
| 09:21.06 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 09:21.10 | MmmmToop | gimp is great... |
| 09:21.16 | MmmmToop | I use it quite a bit |
| 09:21.17 | dudes | yes it is |
| 09:21.32 | MmmmToop | not good with big images though...but otherwise great! |
| 09:21.34 | MustangMatt | *ducking in corner* I still can't get used to the interface. |
| 09:21.47 | hellop | tzafrir_laptop, dunno |
| 09:21.51 | tzafrir_laptop | gimp is great for composing images and such. for simple screenshots, conversions and such it is a slow overkill |
| 09:22.28 | hellop | tzafrir_laptop, come to think of it, I did notice quite a few millisecond lag on that 112/52 pixel image... |
| 09:22.46 | hellop | <giggle> |
| 09:23.06 | MmmmToop | ha ha ; ) |
| 09:23.34 | tzafrir_laptop | hellop, the startup time. Unless you're an artist with gimpl constantly running |
| 09:23.41 | hellop | I do know however, that the Polycom 500 is well worth the money compared to a Budgetone. |
| 09:24.17 | tzafrir_laptop | hellop, I suppose you're one of those calling them bugtones? |
| 09:25.05 | hellop | However, sadly, I can imagine that many customers would have a hard time with the small buttons and could not even read the "Dial" on the screen... |
| 09:25.24 | hellop | These are the customers that I get to sell 21" monitors to. |
| 09:25.36 | kyoo | I'm trying to understand digium hardware... I can today buy a TDM01B and later convert it to a TDM03B or a TDM22B myself by buying modules? |
| 09:25.50 | syle | yes kyoo |
| 09:26.54 | dudes | kyoo - you shutdown your computer ... put a new module in ... turn on your computer ... configure your zaptel.conf for the new modules ... rum ztcfg -vv (no errors, good.) ... run asterisk and away you go\ |
| 09:27.15 | kyoo | syle: dudes: Thank you. |
| 09:27.40 | kyoo | thevoice.digium.com is a brilliant service. :) |
| 09:28.30 | Lan16`spdy^gTp | why odbc dont request my database however odbc is connected to :^o, someone can help me ? |
| 09:28.43 | Lan16`spdy^gTp | when i make odbc show |
| 09:28.55 | Lan16`spdy^gTp | name : connect1 |
| 09:29.00 | Lan16`spdy^gTp | dsn : asterisk |
| 09:29.04 | Lan16`spdy^gTp | connected : YES |
| 09:29.11 | hellop | The budgetone is nice for old people.. big buttons and a nice big LCD. |
| 09:29.35 | Lan16`spdy^gTp | and when i want to make registration it says no registration for peer |
| 09:30.29 | *** join/#asterisk Maksim (n=max@213.142.207.20) |
| 09:30.47 | juraj | kyoo: indeed, if it also worked for something different than english :) |
| 09:31.14 | kyoo | What is a good "multi line" sip phone? One you'd want to put on a receptionist's desk. (dependable, efficient, easy call control, etc...) |
| 09:31.32 | *** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 09:31.33 | drray | cisco 7960 |
| 09:31.47 | X-Rob | Snom 220 or 3600 with extension pad? |
| 09:31.48 | *** join/#asterisk ti4 (n=tia_@82.119.194.149) |
| 09:32.35 | X-Rob | Easy call control would be the GXP-2000, and for multi-line you'd need better firmware (does not exist yet) and the extension keypad (again, does not exist yet) |
| 09:32.54 | X-Rob | gxp has a big 'hold' button. Receptionists like that. |
| 09:33.03 | X-Rob | the snom's is little |
| 09:33.15 | kyoo | But it only supports a single person at once? |
| 09:33.22 | drray | with the 7960 you can forward and hold and confrence, and it has the headset |
| 09:33.36 | X-Rob | drray - so does pretty much every phone now. |
| 09:35.20 | kyoo | gxp-2000 has two switching rj45 ports? So it acts as a switch/hub? |
| 09:35.45 | X-Rob | yep |
| 09:35.58 | X-Rob | again, most phones have that these days. |
| 09:36.08 | kyoo | Convenient, I suppose - I with it would mention if they are 10 or 100. :) |
| 09:36.22 | X-Rob | GXP-2000's are 100, PA1688 based phones are 10. Snoms are 100 |
| 09:37.31 | kyoo | But the gxp-2000 can only handle a singe call at once? Or a I thinking about this wrong, and that all takes place at the PBX? |
| 09:38.31 | kyoo | Can a person with a gxp-2000 take call, see another call coming through, put a on hold, talk to b and pick a back up when he's done? |
| 09:38.47 | Lan16`spdy^gTp | thx for help |
| 09:38.49 | Lan16`spdy^gTp | ... |
| 09:41.25 | *** join/#asterisk zigman (n=zigman@irc.zigman.de) |
| 09:45.56 | *** join/#asterisk nagl (n=nagl@137.208.4.164) |
| 09:47.59 | kyoo | SIP phones with asterisk can be made "as useful" as other PBX phones these days? IE, you can see what lines are in use and available, lines can be parked, held, passed among phones, possibly listened in to by a manager? |
| 09:48.23 | lters | basically yes |
| 09:48.43 | kyoo | "basically" ? :) |
| 09:48.44 | MustangMatt | For anyone interested my Asterisk/debian answering machine guide is maturing: http://kaatman.com/asteriskansweringmachine.html |
| 09:48.44 | MustangMatt | I'd still love to hear any feedback whether or not it's a relatively easy read, etc. |
| 09:48.47 | lters | seeing via the phone, maybe a prob |
| 09:49.09 | hellop | yeah, seeing which lines are in use on the fone would be nice... |
| 09:49.16 | lters | but u can use FOP to see every thing |
| 09:49.18 | *** join/#asterisk Tili[Dinner] (i=Tili@61.144.21.227) |
| 09:49.21 | kyoo | lters: Can you explain a little more, please? |
| 09:49.24 | kyoo | FOP? |
| 09:49.34 | PakiPenguin | hey Tili[Dinner] :p |
| 09:49.35 | hellop | kyoo, you can use a web page to see whats in use |
| 09:50.10 | lters | sip phones can park, hold, transfer, listen in, pickup and all that good stuff. |
| 09:50.29 | lters | but to *see* who is on the phone u need the web piece. |
| 09:50.42 | kyoo | hellop: Ok... Is that because of a missing chunk of technology in asterisk, or missing parts of the SIP spec? |
| 09:51.15 | hellop | lters, what about writing a program for something like the Polycom 500 to see what lines are in use.. |
| 09:51.35 | lters | polycom already can see it via the hint system in * |
| 09:51.38 | kyoo | hellop: Exactly what I was wondering... the phones have the capability? |
| 09:51.54 | lters | some do, some don't |
| 09:52.07 | lters | cisco / sip do not have the firmware support, yet |
| 09:52.20 | lters | cisco / sccp does. and it works now. |
| 09:52.25 | hellop | kyoo, I also would like to know how to write a program for my phone, but yes, you can d/l new programs to it. |
| 09:52.39 | kyoo | So if i buy a polycom 500, then the "lights" will "work"? |
| 09:52.56 | hellop | kyoo no |
| 09:53.05 | X-Rob | hellop - yes, they well. |
| 09:53.12 | X-Rob | They _do_ work. just not as well as you'd expect. |
| 09:53.23 | X-Rob | There's no support for ringing indication, or for call pickup currently |
| 09:53.30 | lters | search for hint and polycom on the wiki. |
| 09:53.34 | hellop | hmm I'll have to look into that. what I search wiki for? |
| 09:53.36 | X-Rob | See mantis bug 3644 for the patch to support all that |
| 09:53.52 | lters | how to set it up. |
| 09:54.02 | X-Rob | lters - if you're using Asterisk@Home it'll just work. |
| 09:54.20 | X-Rob | but I suggest the Snom phones rather than the polycom phones. |
| 09:54.21 | kyoo | X-Rob: But it's all *possible* and probably on it's way sometime? (Faster if I choose to roll up my sleeves?) |
| 09:54.26 | lters | X-Rob, does sip/hints work for making and recieving calls. |
| 09:54.39 | X-Rob | kyoo/lters - as I said, bug 3644 _implements it now_ |
| 09:54.41 | X-Rob | it's just not in CVS |
| 09:54.48 | X-Rob | it's still aimed to go into 1.2 |
| 09:55.05 | kyoo | X-Rob: Ah, "bug" made me think it was a planned feature, not an implemented one. :) |
| 09:55.24 | X-Rob | it's a bug ("it doesn't work") with a patch ("here you go") |
| 09:55.37 | kyoo | X-Rob: Do the snom phones "light up" properly with A@H? Or only the polycom? (going to search "hint" in wiki now...) |
| 09:55.39 | lters | X-Rob, fancy. Using sccp with sccp phones, ie, one sccp monitoring another u even can get callerid... |
| 09:56.01 | lters | kyoo, warning. u may not like the snom phone itself. |
| 09:56.10 | *** join/#asterisk meppl (n=mephisto@84.245.164.92) |
| 09:56.17 | lters | kyoo, I got one and it ended up on the shelf :( |
| 09:56.23 | X-Rob | kyoo - yes, they do. There's an issue or two still to get teased out of it, but it does work. Unfortunately, every fourth or fifth transfer from a snom crashes the asterisk server, but I'm working on that 8) |
| 09:56.35 | X-Rob | lters - Send it to me! |
| 09:56.40 | *** join/#asterisk Tili[Dinner] (i=Tili@61.144.21.227) |
| 09:56.52 | lters | X-Rob, yeah, I should. got the side car and all :) |
| 09:56.52 | X-Rob | http://www.aussievoip.com.au/wiki-Rob+Thomas |
| 09:56.54 | kyoo | lters: What was wrong with it? Did it feel cheap? Slow response? Fragile connection? |
| 09:57.13 | lters | buttons did not feel nice. |
| 09:57.24 | X-Rob | lters - I'll even pay for postage. Seriously 8) |
| 09:57.35 | lters | handset was not at all like the ciscos. ( we were spoiled ) |
| 09:58.14 | *** join/#asterisk RoyK (n=roy@213.160.242.93) |
| 09:58.26 | lters | basically it just felt cheap. the lights were a pail orange instead of red or grenn. |
| 09:58.39 | lters | u bairly can see them light up. |
| 09:58.52 | RoyK | ~seen cypromis |
| 09:58.53 | jbot | cypromis is currently on #asterisk-doc (1d 3h 52m 37s) #asterisk (1d 3h 52m 37s). Has said a total of 21 messages. Is idling for 21h 4m 38s |
| 09:58.55 | RoyK | ~seen the_light |
| 09:58.55 | jbot | RoyK: i haven't seen 'the_light' |
| 09:59.05 | lters | I do like that snom is * friendly |
| 09:59.06 | X-Rob | ~the band |
| 09:59.14 | lters | RoyK, ki |
| 09:59.22 | RoyK | lters: gi |
| 09:59.31 | lters | RoyK, got my first sangoma :) |
| 10:00.27 | kyoo | Neat, a phone accessory with it's own web interface. :) (snom 360) |
| 10:00.43 | juraj | kyoo: i believe every phone has it's web interface |
| 10:00.47 | juraj | voip phone |
| 10:00.54 | kyoo | Oh, I guess i'm just new. :) |
| 10:01.02 | lters | and the later fw really made em better. |
| 10:01.14 | lters | X-Rob, u can msg me. |
| 10:02.33 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:02.41 | hellop | MustangMatt, your setup looks nice and simple.. good job. |
| 10:03.16 | MustangMatt | hellop: Thank you. It's not quite finished. Still need more info in the actual recording messages section but I'm learning as I write it :) |
| 10:03.36 | *** part/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121) |
| 10:05.27 | hellop | lters, I'm not seeing this hint stuff yet.. I searched for "Polycom hint" and "hint polycom line in use" |
| 10:05.51 | kyoo | http://www.voip-info.org/tiki-index.php?page=Asterisk+presence |
| 10:06.49 | *** join/#asterisk nroej (n=joern@lak-39-56.wohnheime.ruhr-uni-bochum.de) |
| 10:07.22 | kyoo | Does "hands free" mean speakerphone? :) |
| 10:07.33 | juraj | aaah, my wiki page :) |
| 10:08.05 | kyoo | juraj: I noticed it was yours. :) |
| 10:08.44 | hellop | loading.... Bueler... |
| 10:08.52 | kyoo | The 320 does not support the sidecar, right? |
| 10:09.15 | kyoo | Or is the sidecar completely standalone? |
| 10:09.25 | kyoo | (snom 360 sidecar) |
| 10:11.18 | nroej | hi all |
| 10:12.18 | kyoo | The aastra 480i' slights are *not* supported by asterisk hinting, correct? |
| 10:12.33 | hellop | Well I checked out presence, isn't that for telling if a certain phone is in use, not for telling if a certain external line is in use? |
| 10:13.47 | *** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 10:15.00 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 10:15.54 | *** join/#asterisk Mc_Tr (n=Mc_Tr@bacterio.knet.es) |
| 10:15.56 | kyoo | Incidentally, if a web page can tell me this information, then I could write an application that can also "get at" this info, correct? |
| 10:16.06 | kyoo | (without scraping the web site.) |
| 10:16.17 | Mc_Tr | hi! |
| 10:16.59 | Mc_Tr | has anybody SPA-841 and asterisk@home working? |
| 10:17.46 | Mc_Tr | my problem is that Sipura SPA-841 don't register in asterisk. |
| 10:18.08 | Mc_Tr | but with same user/pass in X-TEN all works perfectly. |
| 10:18.11 | hellop | kyoo, yes |
| 10:18.35 | jontow | mc_tr; i've got no problems with my SPA-841 registering.. |
| 10:18.35 | Mc_Tr | i apreciate yours comments ;) |
| 10:19.11 | Mc_Tr | jontow, what you configure?, User/pass and proxy sip ? |
| 10:19.28 | jontow | outbound proxy as well.. |
| 10:19.33 | jontow | and make sure you check 'use outbound proxy' |
| 10:19.50 | Mc_Tr | and what outbound proxy use? |
| 10:20.03 | jontow | your * box's host/ip |
| 10:20.25 | Mc_Tr | ok, go to try it. |
| 10:21.07 | jontow | if you can't get it..lemme know, i'll plug my phone in before i leave for work and i'll duplicate the settings for you :) |
| 10:21.20 | Mimmus | good morning, does anyone know the exact meaning of overlapdial=yes in zapata.conf? |
| 10:21.23 | Mc_Tr | ok, thanks jontow |
| 10:21.34 | Madkiss | GNAR. |
| 10:21.47 | *** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121) |
| 10:21.59 | Madkiss | My asterisk still terminates SIP VoIP-sessions for no obvious reason |
| 10:22.07 | *** join/#asterisk andrebarbosa (n=andrebar@adsl-f49-s197-critical-coi.nortenet.pt) |
| 10:22.11 | Mc_Tr | jontow.... don't work :( |
| 10:22.12 | PakiPenguin | Madkiss, your * hates you |
| 10:22.34 | jontow | alright, i've gotta be to work in about 30mins |
| 10:22.44 | Mc_Tr | ok |
| 10:22.44 | jontow | so i'll plug it in before i walk out the door and get you the settings then.. that ok? :) |
| 10:22.48 | Madkiss | == Spawn extension (default, 00216xxxxxxxx, 2) exited non-zero on 'SIP/2000-2bc8' |
| 10:23.06 | Mc_Tr | i'm still waiting for you ;) |
| 10:23.13 | Madkiss | Is there some way to make it tell me more about that, *without* enabling SIP debugging, which makes debugging actually impossible because it prints all kinds of useless messages to the screen? |
| 10:23.46 | tzafrir_laptop | Madkiss, enable debugging and edit logger.conf to not send debugging to the console |
| 10:24.13 | dudes | Madkiss - as useless as it is ... it helps debug the problem |
| 10:24.52 | dudes | Madkiss - But I don't know your issue so I won't pretend |
| 10:24.54 | tzafrir_laptop | dudes, Madkiss , actually the trace from 'verbose' is normally quite useful. The one from 'debug' is often too verbose. |
| 10:25.31 | dudes | tzafrir_laptop - I never said anything about enabling verbose in logger.conf |
| 10:26.08 | tzafrir_laptop | IIRC, the default logger.conf doesn't send 'debug' to the CLI, which is indeed sane. |
| 10:26.22 | Madkiss | Interestingly enough, while calling my own phone and hanging up then, the 7960 will need ~60 seconds to notice that the call is finished |
| 10:27.02 | tzafrir_laptop | Madkiss, the 7960 or asterisk? what does 'show channels' show? |
| 10:27.19 | andrebarbosa | anyone with experience with TE410P installation? |
| 10:27.20 | andrebarbosa | :s |
| 10:27.59 | Mc_Tr | jontow; I solved MY problem..... |
| 10:28.07 | Mc_Tr | in extension net=yes |
| 10:28.15 | Mc_Tr | sorry, nat=yes and work prefectly |
| 10:28.17 | Madkiss | In verbose, I see |
| 10:28.22 | Madkiss | Attempting native bridge of SIP/2000-aebc and SIP/tonline-eade |
| 10:28.25 | jontow | :))) |
| 10:28.30 | jontow | cool |
| 10:28.31 | Madkiss | == Spawn extension (default, 00xxxxxxxxx, 2) exited non-zero on 'SIP/2000-aebc' |
| 10:28.36 | dudes | andrebarbosa - I've done installs with 4 te410P's in one box |
| 10:28.36 | Madkiss | and that's it |
| 10:29.05 | tzafrir_laptop | Madkiss, set verbose 3 |
| 10:29.12 | tzafrir_laptop | to see what's going on |
| 10:29.26 | Madkiss | Verbosity was 38 and is now 3 |
| 10:29.45 | tzafrir_laptop | hmmm..... so this wasn't it, then |
| 10:30.27 | andrebarbosa | dudes, i'm witha strange problem for me.. i do: modprobe wct4xxp |
| 10:30.30 | andrebarbosa | and nothing happens |
| 10:30.32 | Mc_Tr | jontow, i'm goging to test to call |
| 10:30.40 | andrebarbosa | no messages on dmesg, no error messages |
| 10:30.41 | andrebarbosa | :s |
| 10:30.45 | tzafrir_laptop | andrebarbosa, aparantly the module was loaded |
| 10:30.56 | tzafrir_laptop | andrebarbosa, lsmod | grep zaptel |
| 10:30.58 | andrebarbosa | the card is plugin with knigthrider lights |
| 10:31.05 | dudes | andrebarbosa - do a lsmod |
| 10:31.19 | andrebarbosa | zaptel 229316 3 wct4xxp,wcfxs,ztdummy |
| 10:31.29 | andrebarbosa | ok no? |
| 10:31.33 | dudes | if see if are c it ok |
| 10:31.39 | tzafrir_laptop | already loaded. rmmod ztdummy while you're at it |
| 10:31.47 | frenzy | Is this theory possible; Two Asterisks boxes, BoxA & BoxB all sip-to-sip (in-network calls) to take place on BoxB while all other incoming and outgoing to take place on BoxA ? |
| 10:32.08 | tzafrir_laptop | andrebarbosa, there's no point in the extra overhead of the interrupts of ztdummy |
| 10:32.22 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 10:32.34 | *** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2) |
| 10:32.43 | andrebarbosa | ok i removed it, i have it cause the bri cards |
| 10:33.29 | frenzy | any thoughts ? |
| 10:33.32 | Mc_Tr | jontow; are yor here? |
| 10:34.07 | andrebarbosa | but, so now i have the module loadule but no messages on the dmesg and knightrider lights? |
| 10:34.18 | andrebarbosa | this is normal? |
| 10:34.21 | Mc_Tr | i can't to do calls, first "407 Proxy Authentication Required" and second "488 Not Acceptable here" |
| 10:34.35 | Mc_Tr | but my extension it's already registered |
| 10:34.38 | tzafrir_laptop | frenzy, sure. But are there any other limitations you want ot tell us about? (budget limitations?) |
| 10:35.15 | tzafrir_laptop | andrebarbosa, modprobe will simply do nothing if the module has already been loaded. |
| 10:35.30 | opus_ | when asterisk timesout on a registry, can I make it reregister again? |
| 10:35.51 | opus_ | like today, the network went down. and not all lines immediately re-registered |
| 10:36.19 | andrebarbosa | but i read somewhere, that when the module is loaded the lights stop |
| 10:40.25 | *** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 10:40.38 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 10:40.49 | *** join/#asterisk nagl (n=nagl@137.208.4.183) |
| 10:41.10 | *** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3985244.sympatico.ca) |
| 10:41.21 | *** join/#asterisk Tili (n=Tili@61.144.21.227) |
| 10:44.10 | *** part/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg) |
| 10:54.03 | *** join/#asterisk Tili (n=Tili@61.144.21.227) |
| 10:54.15 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 10:55.21 | *** join/#asterisk PakiPenguin_ (i=uppal@unaffiliated/pakipenguin) |
| 10:56.14 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
| 10:56.40 | nicox | hello, do anybody know if there is avariable in the dialplan where overlap digits are saved? |
| 10:58.29 | *** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121) |
| 11:00.48 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 11:00.50 | surfdue | hi |
| 11:01.07 | surfdue | what extention is the defualt menu calling from an internal SIP phone? |
| 11:01.09 | nicox | hi |
| 11:02.16 | surfdue | does anyone know? |
| 11:02.41 | nroej | surfdue: i dont understand your question ;-) |
| 11:02.48 | Jzalae | surfdue: give it half an hour for someone who knows to check this window before you give up |
| 11:03.13 | Jzalae | (i've seen it take an hour and twenty) |
| 11:03.33 | Jzalae | ... and that is normal for this type of channel |
| 11:03.44 | kyoo | What is a consumer-level firewall that supports QOS? |
| 11:03.54 | kyoo | (as it applies to AIX and SIP) |
| 11:04.59 | surfdue | what extention is the defualt menu calling from an internal SIP phone, we made a menu for asterisk voice menu |
| 11:05.06 | surfdue | what is the defualt extention to reach it? |
| 11:05.22 | surfdue | if that helps clear it up some |
| 11:05.30 | nroej | the exten you gave to it?! |
| 11:06.04 | Inv_arp | kyoo: consumer lever firewalls?... never heard of any watchdog isnt too expensive or justs set up a linux box |
| 11:06.07 | surfdue | ahh |
| 11:06.43 | nicox | <PROTECTED> |
| 11:07.19 | X-Rob | nicox - define 'overlap digits' - DTMF signals sent/received after the call is established? |
| 11:07.28 | kyoo | Inv_arp: Would you put asterisk and your firewall on the sam box? Otherwise the noise, hardware cost, maintenance cost, etc doesn't seem worth it... But together is more exploit-prone... |
| 11:07.30 | *** join/#asterisk Praktikant01 (n=lars@dsl-084-059-141-043.arcor-ip.net) |
| 11:07.50 | Praktikant01 | good day |
| 11:07.54 | X-Rob | bad night |
| 11:08.27 | X-Rob | heh |
| 11:09.22 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 11:09.48 | X-Rob | nicox - talk in channel |
| 11:09.51 | X-Rob | not privately |
| 11:10.10 | Inv_arp | kyoo: actually my firewall (iptables) and asterisk is on the same box... for my office (5 sip phones) (2 pots) |
| 11:10.15 | nicox | no, digits recieved before the call established, my problem is, i have the extension 012798357 and i will forward it to zap/g1/016621735202 but the switch of the telco gets from my asterisk 016621735202012798357 why he is doing this? |
| 11:10.41 | X-Rob | Because that's what your dialplan is telling it to do. |
| 11:10.52 | *** join/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121) |
| 11:10.57 | X-Rob | why don't you paste the relevant bits of your dialplan to pastebin? |
| 11:11.07 | surfdue | is there a way to login to amp? |
| 11:11.08 | nicox | no, i tell him to dial only the 016621735202 |
| 11:11.10 | X-Rob | ~pastebin |
| 11:11.10 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://bzflag.pastebin.ca/ |
| 11:11.13 | surfdue | like me im not on the server |
| 11:11.17 | surfdue | can i? |
| 11:11.43 | X-Rob | nicox - I bet you're not. Paste your extensions.conf to pastebin |
| 11:12.01 | surfdue | what port does it use? |
| 11:12.12 | X-Rob | surfdue - 80. |
| 11:12.21 | surfdue | port 80 is web? |
| 11:12.24 | X-Rob | yes |
| 11:12.26 | X-Rob | amp is web |
| 11:12.34 | X-Rob | therefore, amp = 80. |
| 11:12.38 | PoWeRKiLL | hi |
| 11:13.14 | PoWeRKiLL | someone know how to chaange sipura dial plan to allow *21*xxxxx# for call transfer ? |
| 11:13.19 | surfdue | oh i get it |
| 11:13.20 | surfdue | :P |
| 11:13.24 | surfdue | kthx |
| 11:13.37 | tzafrir_laptop | jbot, pastebin is s/bzflag.// |
| 11:13.37 | jbot | ...but pastebin is already something else... |
| 11:15.20 | nicox | http://pastebin.com/340642 |
| 11:15.55 | tzafrir_laptop | jbot, pastebin =~ s/bzflag.// |
| 11:15.55 | jbot | tzafrir_laptop: OK |
| 11:16.11 | tzafrir_laptop | ~patebin |
| 11:16.11 | jbot | i heard patebin is your friend |
| 11:16.31 | tzafrir_laptop | ~pb |
| 11:16.31 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 11:19.11 | syle | get a real job jbot! |
| 11:19.12 | X-Rob | Hrm. OK. That's wierd. |
| 11:19.29 | X-Rob | exten => 0127983,5,dial(zap/g1/1099016621735202) |
| 11:19.37 | X-Rob | <nicox> no, i tell him to dial only the 016621735202 |
| 11:19.41 | X-Rob | So who's retarded here? |
| 11:20.18 | syle | people who can;t code? |
| 11:20.20 | X-Rob | Ah, no, you did mention the 10990 bit earlier |
| 11:20.33 | X-Rob | syle - No, my accountant wife isn't retarded, and she can't code. |
| 11:20.41 | X-Rob | she can count beans better than anyone I know |
| 11:21.08 | tuxinator_linuxM | bean counting, sounds fun ;-) |
| 11:21.24 | syle | shes a woman she has an excuse lol |
| 11:22.02 | syle | if they spread their legs and cook you food what more do you need |
| 11:22.10 | tuxinator_linuxM | Sun is going to be up soon, maybe I should get some sleep? |
| 11:22.14 | riksta | bean flicking more fun |
| 11:22.25 | tuxinator_linuxM | syle: Not married, are you? |
| 11:22.44 | nicox | yes, but the 1099 i need for the telco |
| 11:22.45 | syle | actually i am lol |
| 11:22.52 | X-Rob | I cannot believe I don't have a functional DVD rom here. |
| 11:23.45 | nicox | but the 1099 is okay, the problem is the telco get 0166217352020127983 |
| 11:23.51 | tuxinator_linuxM | Most of my stuff is packed, the wife and I are moving to California next week. (I was looking for me external DVD burner) |
| 11:24.17 | syle | get a dvd 9 |
| 11:25.17 | tuxinator_linuxM | syle: It's a http://www.plextor.com/english/products/716uf.htm |
| 11:26.22 | syle | idk , depends what your budget is , personally i only go with sony for dvdrom,cdrom parts |
| 11:26.23 | X-Rob | nicox - ok, that's just wierd. How are you connecting, and how do you know the telco's getting it? |
| 11:26.46 | tuxinator_linuxM | syle: never had a sony, good stuff? |
| 11:27.16 | X-Rob | Gah |
| 11:27.19 | syle | yeah man, sony makes their shit good, i;ve had alot of my sony stuff for 10 years and still going |
| 11:27.20 | nicox | i see it in the CDR's and i'm connected with 2 E1 one incomming one outgoing |
| 11:27.21 | X-Rob | I just looked _on top_ of my PC |
| 11:27.25 | X-Rob | what's there? A DVD Rom. |
| 11:27.26 | X-Rob | *sigh* |
| 11:28.08 | syle | but only for dvd, cd , vcr's and dvd players |
| 11:28.30 | tuxinator_linuxM | X-Rob: You need a cookie! |
| 11:28.44 | tuxinator_linuxM | X-Rob: or I need a cookie |
| 11:28.49 | X-Rob | nicox - OK. Paste you /etc/zaptel.conf and your /etc/asterisk/zapata.conf to pastebin |
| 11:28.50 | X-Rob | ~pb |
| 11:28.50 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 11:28.51 | X-Rob | 8) |
| 11:29.02 | syle | i think the dvd9;s are still expensive |
| 11:29.04 | X-Rob | tuxinator_linuxM - don't you need to be in bed or something? 8) |
| 11:29.14 | syle | of course prob only sony making it again |
| 11:29.19 | tuxinator_linuxM | X-Rob: Aye, that I do |
| 11:29.35 | tuxinator_linuxM | X-Rob: Wife is sleeping next to me |
| 11:29.51 | X-Rob | Ooh Ooh |
| 11:30.00 | X-Rob | You know what's fun with _pregnant_ sleeping wifes? |
| 11:30.08 | X-Rob | wake up blobby by poking it a couple of times |
| 11:30.09 | tuxinator_linuxM | X-Rob: Cow tipping |
| 11:30.14 | X-Rob | it'll wriggle around and wake her up |
| 11:30.21 | X-Rob | Chicks dig it. Really they do. |
| 11:32.08 | nicox | http://pastebin.com/340651 |
| 11:32.51 | *** join/#asterisk nagl (n=nagl@137.208.4.169) |
| 11:33.09 | X-Rob | nicox - the 'channel =>' is the 'end' of the section. |
| 11:33.20 | X-Rob | so you've got 'overlapdial=yes' only applied to group=3 |
| 11:33.24 | X-Rob | oh |
| 11:33.25 | X-Rob | doh |
| 11:33.33 | X-Rob | sorry, didn't see the line above group=1 |
| 11:34.11 | ful|work | does asterisk and openssi works nicely? |
| 11:34.13 | nicox | its general, overlapdial=yes is for all channels |
| 11:34.18 | X-Rob | if I was you, I'd be messing with the 'pridialplan' and 'switchtype' options myself. |
| 11:34.27 | X-Rob | note, you can shrink that down to: |
| 11:34.31 | X-Rob | group=1 |
| 11:34.40 | X-Rob | signalling=pri_net |
| 11:34.47 | X-Rob | channel =>32-46,48-62 |
| 11:34.55 | X-Rob | group=3 |
| 11:34.59 | X-Rob | signalling (etc etc) |
| 11:35.03 | nicox | http://pastebin.com/340652 |
| 11:35.08 | *** join/#asterisk folsson (n=filip@h82n1fls32o985.telia.com) |
| 11:35.10 | X-Rob | becuase everything inherits from above |
| 11:35.54 | X-Rob | switchtype=euroisdn would be a good start |
| 11:36.00 | *** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net) |
| 11:36.07 | nicox | the configuration i think is okay, so pri_net is working fine and also overlapdial is working, but my problem.... |
| 11:36.11 | X-Rob | (being that you're on european isdn) |
| 11:36.19 | X-Rob | no, the configuration is wrong. |
| 11:36.26 | nicox | no, with switchtype=euroisdn the connection is not working. |
| 11:36.44 | nicox | believe, i have traffic on the lines.... |
| 11:36.50 | nicox | so it must be working |
| 11:37.11 | nicox | and with switchtype=euroisdn i get no link to the telco |
| 11:37.12 | X-Rob | the protocol isn't right |
| 11:37.34 | X-Rob | I'd ask on asterisk-users@lists.digium.com |
| 11:37.58 | nicox | okay thanks, i will do so |
| 11:38.03 | X-Rob | explain your problem, say you're from germany, mention your ISDN provider, explain what's going on, say 'help' |
| 11:38.19 | X-Rob | post links to the pastebin.com dumps of your config files |
| 11:38.29 | nicox | okay, i will do so thanks |
| 11:39.22 | tuxinator_linuxM | I woke up my wife when I got into the cookies. It's all X-Rob's fault. He made me want a cookie. |
| 11:39.47 | *** join/#asterisk r0m (n=SysOp@bl8-28-55.dsl.telepac.pt) |
| 11:40.45 | X-Rob | Eek! |
| 11:41.13 | X-Rob | Tell her the strange australian geek is sorry. |
| 11:42.40 | tuxinator_linuxM | She is a very forgiving wife. |
| 11:43.09 | *** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin) |
| 11:43.24 | *** join/#asterisk robbyloblaw (n=robbylob@host-209-50-87-188.dyn.295.ca) |
| 11:43.34 | tuxinator_linuxM | We are early twenty's, we can take the lack of sleep ;-) |
| 11:43.55 | nicox | I have another POroblem, i will compile asterisk-CVS-HEAD but after 20 seconds i only get this message 1000 time or so on http://pastebin.com/340654 |
| 11:44.14 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 11:44.49 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:45.44 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:45.55 | *** part/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 11:45.59 | *** join/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk) |
| 11:47.13 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 11:48.47 | X-Rob | nicox - Don't know about that one. Tried 'rm -rf /usr/src/asterisk' and started afresh? |
| 11:50.53 | nicox | yes i tried but also not working, the myth is that on this machine i compiled it 3 weeks ago and this version is running... |
| 11:51.46 | tuxinator_linuxM | make clean? |
| 11:52.20 | nicox | i made it, but if the problem occures also the make clean has the same problem |
| 11:53.14 | *** part/#asterisk secure75 (n=mic@p549A3EF7.dip0.t-ipconnect.de) |
| 11:54.14 | *** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com) |
| 11:55.04 | faa_ | Where asterisk is looking for pgsql libs when compiling? |
| 11:55.19 | faa_ | If id add cdr_pgsql.so to MODS |
| 11:56.10 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 11:57.58 | *** join/#asterisk Saulgood (n=Saulgood@82.153.164.84) |
| 11:59.08 | tuxinator_linuxM | i.think.napoleon.dynamiteblows.com .... interesting |
| 11:59.49 | tuxinator_linuxM | Night guys |
| 12:07.31 | nicox | I have ar Problem, i will compile asterisk-CVS-HEAD but after 20 seconds i only get this message 1000 time or so on http://pastebin.com/340654 |
| 12:09.32 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:09.45 | kyoo | Is the flash call manager the only one out there? |
| 12:10.12 | drray | kyoo - gastman |
| 12:11.02 | kyoo | drray: Well, that doesn't really appear to be an improvement. :) |
| 12:11.07 | drray | yes |
| 12:12.16 | kyoo | Is OSX popular with the asterisk community? Would an Aqua tool be appreciated? |
| 12:13.19 | drray | I'm sure it would be appreciated |
| 12:15.26 | macTijn | kyoo: that would be appreciated. What kind of tool did you have in mind ? |
| 12:16.20 | kyoo | macTijn: I'm not sure, I'm noodling design right now. Something with a heavy emphasis on speed of operation and usability. |
| 12:16.23 | *** join/#asterisk cliffton_2005 (i=cliffton@adsl-69-105-116-203.dsl.pltn13.pacbell.net) |
| 12:16.51 | drray | I've been kinda (ie, not at all) working on a gkrellm plugin for asterisk |
| 12:18.00 | *** join/#asterisk klapzin (n=klap@201-0-66-216.dsl.telesp.net.br) |
| 12:18.54 | klapzin | where i can view a modem list ( generic ) whith asterisk suport? |
| 12:19.23 | *** join/#asterisk ttyp0 (n=ttyp0@83.Red-83-55-35.pooles.rima-tde.net) |
| 12:21.52 | drray | ~wiki |
| 12:22.00 | *** join/#asterisk trasschaert (n=karl@212.68.197.226.brutele.be) |
| 12:22.11 | trasschaert | Hello everybody |
| 12:22.51 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 12:23.24 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 12:25.22 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 12:26.12 | trasschaert | Did someone can help me? |
| 12:26.24 | nicox | ask and we will se |
| 12:26.27 | nicox | see |
| 12:26.41 | trasschaert | I just install a digium X100P card into my asterisk, i can make outbound call but i cannot receive call |
| 12:26.49 | trasschaert | I use asterisk@home |
| 12:26.54 | opus_ | read the manual on zapta.conf |
| 12:27.12 | trasschaert | if i call 7777, th ephone ring |
| 12:28.55 | cliffton_2005 | Hello Trasshaert, how much is digium X100P card cost? |
| 12:29.02 | *** join/#asterisk zoo (i=nobody@ip-54-16.travedsl.de) |
| 12:29.05 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 12:29.18 | nicox | what happened if the call knocks on the asterisk? |
| 12:29.47 | trasschaert | I've buy a OEM card for only 6,5$ |
| 12:29.52 | nicox | did anybody installed the current asterisk-cvs-head version? |
| 12:30.19 | *** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com) |
| 12:30.21 | trasschaert | it only ring on the calling phone, nothing append on the sip phone |
| 12:30.30 | klapzin | http://www.laptops4me.com/product_info.php/modem/all-56k-modems/p/v-92-pci-intel-w-voice-s-w-modem/cPath/176_239/products_id/1444 |
| 12:32.35 | lathos42 | Good Morning/Afternoon/Evening/Night everyone |
| 12:32.36 | nicox | hm, behind a nat? |
| 12:32.55 | *** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl) |
| 12:33.03 | trasschaert | no |
| 12:33.34 | trasschaert | i can make call from sip to pstn but not from pstn to sip... |
| 12:34.07 | trasschaert | In amp, i configure incoming call to ring to a extension |
| 12:34.10 | nicox | do you see the call in asterisk console? |
| 12:34.23 | trasschaert | how can i check it? |
| 12:34.32 | trasschaert | where can i see it |
| 12:34.40 | nicox | login via ssh |
| 12:34.45 | nicox | type asterisk -r |
| 12:34.52 | nicox | and see what happens |
| 12:35.37 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 12:35.38 | trasschaert | i'v ethe asterisk cli, when a make a call nothing appen |
| 12:37.06 | spdy | hi everybody |
| 12:37.11 | nicox | hi |
| 12:37.17 | nicox | type set verbose 9 |
| 12:37.21 | nicox | and set debug 9 |
| 12:37.37 | spdy | i have a core dump when i try au authenticate a user using mysql |
| 12:37.37 | nicox | if there also nothing happened, the call does not come to the asterisk |
| 12:38.10 | trasschaert | nothing append... |
| 12:38.17 | spdy | someone now this bug ? |
| 12:38.26 | trasschaert | i already put a analog phone on the line and it ring |
| 12:38.34 | spdy | core dump with mysql iax authentification |
| 12:38.39 | trasschaert | so the line is ok, i already exchange the card |
| 12:39.16 | trasschaert | something special to configure in the zapata.conf? |
| 12:39.27 | nicox | paste it and let us see |
| 12:39.47 | kyoo | How often do the digium cards fail? (Is replacement warranty worth it?) |
| 12:40.07 | jontow | kyoo; i've yet to have one fail, but i've heard of others having it happen |
| 12:40.20 | jontow | we always keep 1 spare of everything we buy thats in a critical box |
| 12:40.29 | jontow | ie. for every 1 T100P in service, we have a spare sitting in hotswap box |
| 12:40.37 | spdy | i have a core dump on asterisk when iax try to authenticate a user using realtime mysql |
| 12:40.38 | *** join/#asterisk docelm0 (n=docelm0@67.106.194.90.ptr.us.xo.net) |
| 12:40.42 | spdy | no one knows ? |
| 12:40.52 | jontow | configs and data are tar zcvf'd nightly and then scp'd across a link to the other, where they're untarred |
| 12:40.57 | kyoo | And how necessary is technical support, considering I already have an asterisk server up and working via SIP? |
| 12:41.03 | jontow | so all we have to do is start asterisk and swap cables |
| 12:41.11 | kyoo | Is it more complicated than I expect? |
| 12:41.13 | jontow | its low-cost redundancy |
| 12:41.22 | opus_ | spdy - me too |
| 12:41.24 | jontow | you get free installation support anyway. |
| 12:41.35 | opus_ | spdy - i saw that.. its weird |
| 12:41.45 | jontow | if you need help after the install and this channel can't provide it.. good luck, at that point you're gonna pay for support and it'll be worth it if you're on a deadline :) |
| 12:42.00 | spdy | opus - arf |
| 12:42.03 | jontow | the digium folks know their shit and will get you what you need :) |
| 12:42.05 | tzafrir_laptop | jontow, sounds lame;-) tar czf - | ssh other server 'cat >archive.tgz' |
| 12:42.08 | spdy | opus - you have the last tarball ? |
| 12:42.16 | opus_ | yeah |
| 12:42.19 | opus_ | cvs from yesterday |
| 12:42.33 | opus_ | spdy - i don't even use IAX but have iaxpeer/iaxusers pointed to my sip table |
| 12:42.43 | Beirdo | FUCK |
| 12:42.46 | opus_ | i mean my 'users' table |
| 12:42.50 | tzafrir_laptop | (and my last ssh command was actually highly lame) |
| 12:42.55 | jontow | tzafrir; very.. :) |
| 12:43.02 | jontow | i gave the idea of it.. its actually a pipeline |
| 12:43.03 | Beirdo | something seemingly toasted my 10/100 switch |
| 12:43.06 | opus_ | spdy - the backtrace saids it something with strlen |
| 12:43.13 | Beirdo | now my asterisk box can't reach the world |
| 12:43.15 | spdy | opus - ok |
| 12:43.18 | Beirdo | :( |
| 12:43.29 | jontow | tar zcvf - | ssh otherserver 'tar zxvf -' <-- thats more of the idea ;) |
| 12:43.40 | spdy | opus - i know that sip has got the same probleme one time and they fix it |
| 12:43.40 | opus_ | spdy - nobody seems to want to work on it. are you a programmer? i'd like to work on it |
| 12:44.03 | kyoo | How do the polycom 300 and 301 differ? |
| 12:44.22 | Beirdo | I guess I'll be buying a new one |
| 12:44.24 | Beirdo | POS |
| 12:45.25 | spdy | opus - its a problem in the chan_iax2 ? |
| 12:45.32 | opus_ | no, realtime |
| 12:46.08 | spdy | ok |
| 12:46.10 | spdy | :s |
| 12:46.37 | trasschaert | nicox: sorry, i'm not a linux user, how can i copy all the file? |
| 12:46.44 | trasschaert | whats the vi command? |
| 12:47.10 | nicox | sorry, i don't know asterisk@home so i can't answer |
| 12:47.11 | opus_ | trass you should really learn unix and come back :( |
| 12:47.40 | opus_ | some ppl might get offended by the questions. 'ask a smart question and get a smart reply' is the general guideline |
| 12:47.58 | opus_ | spdy - do you use realtime mysql or realtime odbc? |
| 12:48.05 | nicox | <PROTECTED> |
| 12:48.07 | spdy | opus - realtime mysql |
| 12:48.12 | nicox | can anybody help me? |
| 12:48.21 | opus_ | nicox - yeah, like an infinite loop? |
| 12:48.21 | spdy | opus - with the last tarball version on the digium site |
| 12:48.32 | opus_ | spdy - yeah. did you ever try odbc? |
| 12:48.53 | spdy | opus - odbc is no longer supported in the last tarball |
| 12:48.57 | spdy | opus - :( |
| 12:49.01 | trasschaert | ; Zapata telephony interface |
| 12:49.02 | trasschaert | ; |
| 12:49.02 | trasschaert | ; Configuration file |
| 12:49.02 | trasschaert | [trunkgroups] |
| 12:49.02 | trasschaert | [channels] |
| 12:49.04 | nicox | and why? |
| 12:49.11 | *** join/#asterisk Katty (n=ladykatr@68.112.15.110) |
| 12:49.23 | nicox | not paste here! |
| 12:49.26 | opus_ | nicox - make clean, get the latest source again, make libpri and zapata again. |
| 12:49.38 | opus_ | that seems to fix it but i couldn't figure out why |
| 12:49.38 | nicox | i did so |
| 12:49.48 | nicox | libpri and zaptel is okay |
| 12:50.12 | opus_ | you need to rebuild it. my only guess is that there is a makefile command in /asterisk that checks the date on include files from ../libpri and ../zaptel |
| 12:50.16 | nicox | but if i try to compile asterisk i have this problem |
| 12:50.21 | *** join/#asterisk nagl (n=nagl@137.208.4.182) |
| 12:50.35 | *** join/#asterisk PakiPenguin_ (i=uppal@unaffiliated/pakipenguin) |
| 12:50.35 | nicox | i recompiled it! |
| 12:50.53 | opus_ | spdy - it sucks. odbc isn't supported. i really would like to use it. i hate mysql. |
| 12:50.58 | nicox | but i will try to delete all download from cvs and recompile again |
| 12:51.01 | opus_ | spdy - what do you use for voicemail users? |
| 12:51.11 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 12:51.34 | Katty | morning. |
| 12:51.38 | spdy | opus - now i use static but in a few times i will use mysql |
| 12:52.02 | spdy | opus - yes odbc seem easy to use its idiot :( |
| 12:52.08 | opus_ | spdy - i made the mistake of using mysql for voicemail users.. arge.. i had to patch the code to change the voicemail password |
| 12:52.31 | spdy | opus - :$ |
| 12:52.39 | spdy | opus - not really cool :s |
| 12:53.02 | opus_ | i really really dislike mysql. it corrupts my data |
| 12:53.09 | spdy | opus - i want them to fix the ****** iax core dump |
| 12:53.21 | spdy | opus - i use pgsql sometimes |
| 12:53.40 | *** join/#asterisk morfe (n=morfe@tor/session/x-e805691422953e2d) |
| 12:53.42 | opus_ | do you have valgrind installed? |
| 12:53.53 | spdy | tar xvfz aster..... |
| 12:53.54 | spdy | make |
| 12:53.57 | spdy | make valgrind |
| 12:53.59 | spdy | make install |
| 12:54.00 | spdy | :) |
| 12:54.02 | opus_ | spdy can you load up asterisk with gdb, load the core file, and type 'bt' cp to pastebin i want to see it |
| 12:54.37 | opus_ | http://pastebin.ca/20364 |
| 12:54.52 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 12:54.53 | *** part/#asterisk Eminence (n=Eminence@cpe-24-198-66-186.maine.res.rr.com) |
| 12:54.54 | spdy | no pb |
| 12:54.58 | spdy | i do it now |
| 12:54.59 | spdy | wait |
| 12:55.14 | nicox | compile is running, so i hope now its working |
| 12:55.37 | opus_ | nicox you are the only other person i know thats had this problem. |
| 12:55.46 | opus_ | nicox from a brand new install on a box I get that problem. |
| 12:56.21 | morfe | hello. someone can say me where I can buy a hfc-s card? |
| 12:56.43 | spdy | opus - tu es francais :p ? |
| 12:56.47 | nicox | the problem is there again |
| 12:57.02 | *** join/#asterisk uma62 (n=sudhir@pool-71-114-89-44.washdc.dsl-w.verizon.net) |
| 12:57.07 | opus_ | <--- only knows english and C |
| 12:57.08 | opus_ | :) |
| 12:57.15 | nicox | but, the miracle is that 3 weeks ago i compiled it without this problems |
| 12:57.18 | spdy | opus - ok |
| 12:57.21 | morfe | spdy> moi oui.. |
| 12:57.23 | *** join/#asterisk tuyan (n=tuyan@81.214.255.57) |
| 12:57.26 | spdy | hello morfe |
| 12:57.27 | spdy | :) |
| 12:57.33 | spdy | ca fait du bien des frenchies ici :D |
| 12:57.46 | morfe | spdy> :p |
| 12:57.57 | opus_ | nicox fuck. try this. cd /usr/src ; find ./ -exec touch {} \; |
| 12:59.20 | spdy | :) |
| 12:59.36 | *** join/#asterisk dsfr (n=dsfr@digium.com) |
| 13:00.27 | spdy | opus - gdb display no stack when i load asterisk |
| 13:00.53 | opus_ | not even 'bt'? |
| 13:02.07 | spdy | in shell : gdb asterisk |
| 13:02.11 | *** part/#asterisk B4 (n=B4@202.69.48.245) |
| 13:02.11 | uma62 | I am using a 3 Polycom phones. From time to time, the calls will go to voicemail straight away. I can always make a call from the phones though. Any solution for that |
| 13:02.14 | spdy | (gdb) bt |
| 13:02.19 | *** part/#asterisk the_devil_dont_s (n=the_devi@62.77.178.121) |
| 13:02.19 | spdy | No Stack |
| 13:02.22 | spdy | :s |
| 13:02.30 | opus_ | oh |
| 13:02.36 | opus_ | set your ulmit -c unlmited |
| 13:02.43 | zedkatuf | I have set up a VOIP to PSTN gateway, and I can ring my landline number from my computer, but I can hardly hear myself on my landline..I'm wondering if thr rtp stream might not be getting through.....is there any way to diagnose this at all? |
| 13:02.44 | opus_ | then, gdb> load core.90210 |
| 13:02.56 | opus_ | well, create the core first |
| 13:03.53 | *** join/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com) |
| 13:03.56 | spdy | ok |
| 13:04.09 | funxion | zedkatuf are you using a digium card for pstn gateway? |
| 13:04.24 | DaPrivateer | got disconnected dunno if that went through. if it did im sorry for repeating myself: |
| 13:04.25 | DaPrivateer | question. I am using a Digium X100P; when transferring a call I need to send a hook flash (doing it through an AGI script). ive tried a few different things that did not work. anyone have any ideas how I can do this? |
| 13:05.37 | zedkatuf | funxion: No, I'm using a softphone atm, so in other words I'm doing VOIP ---> PSTN |
| 13:05.52 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 13:06.10 | kyoo | Is it not possible to purchase an unlocked RT31P2 ? |
| 13:07.13 | *** join/#asterisk trasschaert (n=karl@212.68.197.226.brutele.be) |
| 13:07.33 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 13:09.38 | kyoo | Is anyone a service provider that will sell me one with 1 day's service? |
| 13:09.40 | jontow | hmm.. is that the vonage box? |
| 13:09.44 | *** join/#asterisk jiro5281 (n=jiro5281@203.131.137.76) |
| 13:10.01 | kyoo | jontow: It's a linksys router firewall with QOS and two FXS ports. |
| 13:10.09 | jontow | hmm |
| 13:10.12 | Katty | mrow? |
| 13:10.17 | kyoo | And I can't understand why the general public can't buy them... |
| 13:10.27 | jontow | http://support.sipphone.com/index.php?_a=knowledgebase&_j=rate&_i=61&type=yes |
| 13:10.34 | jontow | if thats the box.. |
| 13:10.36 | jontow | then thats your answer |
| 13:10.46 | jontow | i suspect it is not, however. |
| 13:11.14 | kyoo | I don't think so - no clue, actually... |
| 13:11.49 | kyoo | I'm on hold with viopsupply - they are checking if I provide voip service, am I voip service provider... |
| 13:12.02 | *** join/#asterisk epablo (n=epablo@200.75.139.188) |
| 13:12.10 | epablo | Hello People! |
| 13:12.15 | opus_ | <PROTECTED> |
| 13:12.19 | opus_ | http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging |
| 13:13.10 | kyoo | No, but I can buy the exact same product from Draytec for two hundred dollars more.... |
| 13:13.50 | kyoo | WTF is up with the linksys product? I don't understand why I can't buy one - why would linksys care who I am? Do they get kickbacks from vonage or something? |
| 13:14.46 | opus_ | yes from the VOIP mafia |
| 13:15.49 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:16.33 | kyoo | WTH, same problem everywhere I go - why is this device not for sale to the public? |
| 13:16.42 | fenlander | kyoo: the general public can buy them in the uk... |
| 13:16.49 | epablo | I've just pluggin my first E1 on a TE110P. I can get calls to work both ways with iBLC.. but if a use g729.. the is a translation problem |
| 13:17.00 | kyoo | fenlander: I wonder how much the shipping would be. |
| 13:17.01 | *** join/#asterisk DrWho (n=MIKE@mike-new.tc3net.com) |
| 13:17.11 | *** part/#asterisk Maksim (n=max@213.142.207.20) |
| 13:17.22 | olivier_ | <epablo> do you have g729 licences ? |
| 13:17.34 | *** join/#asterisk eminence_ (n=achin@cpe-24-198-66-186.maine.res.rr.com) |
| 13:17.41 | juraj | question about connecting FXO a FXS ports.... if i want to replace a pbx, I can get a digium card with T1/E1 interface. But it actually has only two plugs. What do I need to actually have 50 plugs for ports? :) |
| 13:17.54 | juraj | I'm completely new to telephony, I was just doing voip until now :) |
| 13:18.14 | jontow | juraj; they call 'em a channel bank |
| 13:18.24 | jontow | :) there is much info on the wiki about them |
| 13:18.26 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 13:18.27 | epablo | olivier_: not at this time. I thought that the passthough would work |
| 13:18.31 | DrWho17 | is there a mysql only version of the odbc voicemail storage? |
| 13:18.36 | juraj | jontow: thanks, I'll look at the wiki |
| 13:18.39 | jontow | its pretty much ethernet<-->RJ11*many |
| 13:18.42 | fenlander | kyoo: no idea, but you are probably better off with an spa 2002 |
| 13:19.01 | kyoo | fenlander: Does the spa2002 do firewall, NAT and QOS? |
| 13:19.33 | jontow | no |
| 13:19.37 | jontow | its an ATA |
| 13:19.43 | RoyK | kyoo: not only that, it makes coffee as well |
| 13:19.52 | RoyK | jontow: an ATA can do that as well |
| 13:19.58 | jontow | yeah but not the SPA2002 |
| 13:20.00 | RoyK | jontow: perhaps apart from the coffee |
| 13:20.10 | jontow | if it can.. im interested |
| 13:20.11 | jontow | i've got one :) |
| 13:21.05 | kyoo | the 2002 is basically a 2000 with a 10BT hub built in, right? |
| 13:21.05 | fenlander | :) |
| 13:21.42 | *** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 13:22.04 | RoyK | kyoo: no, it's more like a 2001 |
| 13:22.12 | RoyK | 2000 doesn't have QoS, no T.38 etc |
| 13:22.53 | morfe | someone can say me where I can buy a hfc-s card? |
| 13:23.08 | RoyK | anywhere |
| 13:23.20 | kyoo | So, is anyonein here a service provider that can sell me a RTP31P2-NA? |
| 13:23.51 | *** join/#asterisk BoDePlOt (i=user@pool-68-162-3-185.nwrk.east.verizon.net) |
| 13:23.59 | opus_ | spdy yuck. realtime_peer (peername=0x81134c0 "4\021\b4\021\b") at chan_iax2.c:2593 |
| 13:24.55 | spdy | opus - so whats the problem ? |
| 13:25.15 | opus_ | is your peername in like unicode or something? |
| 13:25.21 | opus_ | thats fucked |
| 13:25.29 | opus_ | i think your problem is different then mine |
| 13:25.54 | spdy | by peername you think var name ? |
| 13:26.02 | opus_ | The value yes |
| 13:26.16 | spdy | 601 |
| 13:26.20 | spdy | its the number |
| 13:26.21 | kyoo | Did vonage underwrite the cost of development on this device? |
| 13:26.39 | spdy | and in realtime mysql name = number |
| 13:27.01 | opus_ | dude something completely fucked realtime |
| 13:27.14 | *** part/#asterisk morfe (n=morfe@tor/session/x-e805691422953e2d) |
| 13:27.16 | spdy | i think so |
| 13:27.35 | opus_ | i'm not at my dev site, but, what if you check out an older version of asterisk / asterisk_addons and rebuild? |
| 13:27.55 | spdy | its what we are going to do |
| 13:28.02 | opus_ | cvs co -D "Aug 1 2005" asterisk asterisk-addons |
| 13:28.22 | spdy | ok |
| 13:28.28 | spdy | i take it |
| 13:28.30 | spdy | and test |
| 13:29.19 | *** part/#asterisk juraj (i=juraj@smtp.rebelium.net) |
| 13:29.42 | opus_ | does asterisk-cvs include changes to asterisk-addons? |
| 13:30.14 | *** join/#asterisk tuyan (n=tuyan@81.214.255.57) |
| 13:32.16 | nick125_lappy | anyone got any links to a cheap ATA device that works with asterisk? |
| 13:32.24 | MmmmToop | exit |
| 13:32.28 | jontow | voipsupply.com has a Sipura SPA-2002 |
| 13:32.35 | jontow | very nice device, i think i paid $69.95 for it |
| 13:32.40 | jontow | 2 FXS ports |
| 13:32.57 | nick125_lappy | how hard to config? |
| 13:33.06 | jontow | its got a myriad of web-config options |
| 13:33.24 | jontow | really, you only need to touch the "User 1" and "User 2" pages, changing.. iirc, 4-5 things per page |
| 13:33.28 | nick125_lappy | 202 has two ports? |
| 13:33.35 | nick125_lappy | 2002 has two ports? |
| 13:33.40 | jontow | (ip address twice, user, pass, checkbox to say use outbound proxy) |
| 13:33.42 | jontow | yes |
| 13:34.44 | nick125_lappy | is there a difference other thenthe one port between the 1001 and the 2002 |
| 13:36.18 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 13:36.19 | nick125_lappy | is the iaxy better for asterisk? |
| 13:37.07 | jontow | the iaxy supports digium |
| 13:37.12 | *** join/#asterisk teddey (i=electro@never.needs.a.reasontodrink.com) |
| 13:37.15 | jontow | its a little more pricy, a little smaller, and much cooler looking |
| 13:37.21 | drray | and it works through nat |
| 13:37.22 | jontow | it does work a bit better being based on IAX2 but it has limitations. |
| 13:37.30 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 13:37.45 | drray | the Iaxy does not have a web configure util |
| 13:37.59 | teddey | hey does anybody know that the theoretical max users/conferences are with meetme? |
| 13:38.01 | jontow | ie. only supports a small subset of codecs, you provision it from the asterisk machine with a command line tool and a text config file (but its easy.) |
| 13:38.13 | jontow | and it doesn't do DNS.. so you have to input the IP address of your * server, not the hostname |
| 13:38.26 | drray | yeah, if you can make head from cvs you can provision an Iaxy |
| 13:38.30 | nick125_lappy | thats usually no problem, because i have to enter the ip anyways |
| 13:38.30 | jontow | agreed.. |
| 13:38.31 | jontow | :) |
| 13:38.39 | jontow | it does ulaw and adpcm, iirc |
| 13:38.46 | jontow | so if you have low-bandwidth links, it may not be for you. |
| 13:38.55 | drray | an Iaxy is a perfect device for use with wrtg54 |
| 13:38.59 | jontow | agreed |
| 13:39.09 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 13:39.11 | drray | you can provision it from the wrt I believe |
| 13:39.18 | nick125_lappy | it wont be used over the internet, just wireless |
| 13:39.22 | jontow | lets just say.. you can tape the thing to a doorjam its so small |
| 13:39.40 | drray | it does get warm, and the blue light is pretty damn bright in my darkened living room |
| 13:39.54 | drray | but it sounds great |
| 13:40.11 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 13:41.00 | teddey | so far with a quad t1 card and a single xenon 3 ghz 2G of ram i can get 96 people on and in conferences the quality is fine |
| 13:41.01 | DaPrivateer | I am using a Digium X100P; when transferring a call I need to send a hook flash (doing it through an AGI script). ive tried a few different things that did not work. anyone have any ideas how I can do this? please? |
| 13:41.03 | drray | if I was going to hand an idiot an ATA and tell him to go home and hook it up by himself, an Iaxy may not be right |
| 13:41.17 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 13:42.07 | drray | I've also not been able to get a fax to go over teh Iaxy |
| 13:42.25 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
| 13:43.16 | nicox | <PROTECTED> |
| 13:44.13 | *** join/#asterisk toddf (n=toddf@net-66-210-104-252.theshop.net) |
| 13:46.29 | *** join/#asterisk nagl (n=nagl@137.208.4.190) |
| 13:47.27 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:48.21 | Ariel_ | hello everyone |
| 13:49.01 | teddey | hey does anybody know that the theoretical max users/conferences with meetme? |
| 13:49.08 | *** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 13:49.26 | Hmmhesays | oh about 5 |
| 13:49.44 | Ariel_ | LOL 5... |
| 13:51.09 | bjohnson | teddey: I think it all depends on hardware power |
| 13:51.15 | Ariel_ | morning Katty |
| 13:51.21 | nicox | <PROTECTED> |
| 13:51.41 | teddey | heh ive already had 96 so i dont believe its 5 |
| 13:51.47 | Ariel_ | the meetme depends on hardware, codec and much more. So there is no real number |
| 13:52.00 | teddey | there is no hard limit is what your saying? |
| 13:52.10 | drray | not built in to the software |
| 13:52.14 | *** join/#asterisk frenzy (n=frenzy@193.220.82.108) |
| 13:52.20 | Ariel_ | teddey, as far as I know no. |
| 13:52.44 | frenzy | Aug 19 13:51:29 WARNING[24526]: chan_sip.c:4832 check_auth: Stale nonce received from '<sip:7777777@xx.xx.xx.xx>' |
| 13:53.14 | teddey | well then you can put it in the books that a xenon 3G with 2M of cache and 2G of ram will do 96 users |
| 13:53.16 | frenzy | what does that mean ? my UA connects after a while I get that |
| 13:53.25 | teddey | when the ds3000p is released ill let you know what the max is |
| 13:53.55 | Ariel_ | teddey, it really depends on the codec as well which one were you using for all the channels |
| 13:54.01 | *** join/#asterisk Hmm-work (i=negative@66.173.103.108) |
| 13:54.13 | Ariel_ | frenzy, check your qualify=60 or less |
| 13:54.57 | teddey | G.711 ulaw all of my calls are from the PSTN |
| 13:55.15 | *** join/#asterisk likwid-- (n=likwid@nc-69-68-67-119.dyn.sprint-hsd.net) |
| 13:55.31 | Ariel_ | teddey, that is why no transcoding. |
| 13:57.26 | kyoo | Does anyone know how to crack one of these RT31P2 routers? If I can't buy one unlocked, can I just unlock one that I've bought? |
| 13:58.21 | opus_ | kyoo |
| 13:58.22 | teddey | why would i want to transcode? |
| 13:58.28 | opus_ | you need to intercept the dhcp request |
| 13:58.47 | opus_ | then change the traffic to download your own rom |
| 13:58.49 | teddey | that just adds more load |
| 13:58.52 | opus_ | EVERYTIME it reboots |
| 13:59.27 | kyoo | opus_: Can't I just make a DNS change? |
| 13:59.41 | Ariel_ | teddey, that is what I mean if you do transcoding it will be less connections. |
| 13:59.48 | Ariel_ | kyoo, no |
| 14:00.18 | kyoo | Hrm.. So stupid - why are they sold like this? Did vonage pay for their development? |
| 14:00.27 | opus_ | yup |
| 14:00.36 | Ariel_ | kyoo, they paid yes lots of money |
| 14:00.44 | teddey | right but i have 96 people dialed in from the pstn transcoding would kill me |
| 14:00.50 | kyoo | I hate corporations... |
| 14:01.05 | opus_ | hey. open source hippy meets wall street |
| 14:01.13 | Ariel_ | teddey, that was my point you asked how many can I get. I posted that it depends on transcoding. |
| 14:01.30 | teddey | ah sorry i didnt understand what you were saying |
| 14:01.31 | kyoo | More like Open Source Hippy tries to beat wall street... |
| 14:02.01 | kyoo | Is there a similar device for a reasonable price? NAT FW, ATA, QOS router |
| 14:02.02 | Katty | yay for hippys |
| 14:02.10 | opus_ | smoke weed everyday |
| 14:03.45 | Ariel_ | Big Biz is the reason we have high oil cost... The Future market is driving the oil cost higher then it should be. (There is no actuall shortage)... |
| 14:04.37 | mut | no |
| 14:04.39 | file[laptop] | blame Canada |
| 14:04.42 | mut | it's all the hippies fault |
| 14:04.50 | file[laptop] | it was in a song, so it must be true |
| 14:05.10 | zedkatuf | newb Question: If I set my softphone up to use a stun server, will that mean it will attempt to bypass my asterisk box? |
| 14:05.45 | olivier_ | stun server just says your softphone your external I |
| 14:05.49 | olivier_ | IP |
| 14:05.52 | mut | i can't believe Zafi.B is still roaming around |
| 14:06.03 | mut | en-mass |
| 14:06.19 | zedkatuf | olivier: does that mean I'd still be going through my asterisk box, though? |
| 14:06.32 | Equinox | Any issues with the grandstream HT-286? |
| 14:06.52 | olivier_ | yes. your soft phone use your external IP in the packet it send asterisk instead of local ip |
| 14:07.03 | Ariel_ | zedkatuf, yes it does unless you use canreinvite=yes |
| 14:07.48 | zedkatuf | olivier_ - ok that's useful to know, tnx |
| 14:08.24 | DaPrivateer | lemme ask a different question. can anyone tell me how, in AGI, to dial on the zaptel channel that is in use? |
| 14:08.53 | zedkatuf | Ariel - tnx..I've been using AMP to set this up, and afaik it doesn't add canreinvite=yes by default (Icould be wrong abt this though!!) |
| 14:10.47 | *** join/#asterisk secure75 (n=mic@ppp-82-135-1-232.mnet-online.de) |
| 14:10.59 | Hmmhesays | no it doesn't |
| 14:11.24 | mover | any one here use cosini ss7 stack? |
| 14:13.17 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 14:13.59 | *** join/#asterisk coppice (n=chatzill@36.199.17.210.dyn.pacific.net.hk) |
| 14:14.19 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
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| 14:20.10 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 14:21.02 | *** join/#asterisk Mensored (n=Barahari@gremesh1.demon.nl) |
| 14:21.07 | kyoo | Doe anyone have an actual phone number for verilan? |
| 14:21.13 | kyoo | s/Doe/Does/ |
| 14:22.37 | ManxPower | <PROTECTED> |
| 14:22.37 | *** join/#asterisk Dublin_drupaller (n=dub@83-70-36-157.b-ras1.prp.dublin.eircom.net) |
| 14:22.43 | ManxPower | that's funny |
| 14:22.52 | Dublin_drupaller | hiya guys... |
| 14:22.56 | |dennis| | Question: We presently have a nortel pbx system that we rent fom a local phone company. We have three pots lines that we use to make calls with and receive calls on. I wish to move to an asterisk system. I intented to use IP Phones. So as far as i can tell all i need is an asterisk box, a 4port fxo for the three + maybe 1 lines and i am good to go. Am i correct?? Please Help... |
| 14:23.21 | DrWho17 | sound fine |
| 14:23.24 | Faithful | anyone know much about the debian version of Zaptel? |
| 14:23.34 | Dublin_drupaller | is this an okay place to ask a (probably stupid) question about asterisk? |
| 14:23.44 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 14:23.49 | Nivex | Faithful: I know a little from when I had to get ztdummy built. What ya need? |
| 14:24.12 | jontow | dublin; we get many of those a day.. just give it a shot |
| 14:24.28 | Faithful | Oh I want to know if the debian version of 1.0.9 has the bristuff patched against it |
| 14:24.33 | nick125_lappy | Dublin_drupaller, i do it all the time, so, you shouldnt have a problem :) |
| 14:24.38 | |dennis| | DrWho17..is there any particular 4port fxo card that you recommend?? I was looking at the Digium TDM04B |
| 14:24.42 | Nivex | Faithful: ah, there I cannot help you. |
| 14:24.43 | tzafrir_laptop | Faithful, any problems? |
| 14:25.15 | tzafrir_laptop | Faithful, http://updates.xorcom.com/iso/ , there's an apt source attached at the end |
| 14:25.17 | DrWho17 | dennis: nope, never used one, I've only used t1's |
| 14:25.34 | Dublin_drupaller | lol..thanks jontow..I'm using drupal (php based CMS) and I was thinking of bolting asterisk onto a site I'm working on...is that a ridiculous idea? Has anyone integrated asterisk into a website community? |
| 14:25.37 | jontow | dennis; that is the one we recommend. |
| 14:25.38 | Faithful | Well I am trying to get zaphfc built against the debian version |
| 14:25.46 | DrWho17 | a didium card is probably going to use the least tinkering to get working properly |
| 14:25.54 | tzafrir_laptop | it's only bristuff RC8h (actually equivalent to RC8j), but should be good enough |
| 14:26.04 | Faithful | they give you the module source no instructions and they don't compile |
| 14:26.06 | nick125_lappy | Dublin_drupaller, you could use asterisk for conferencing, so, it could work like that |
| 14:26.06 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 14:26.09 | |dennis| | jontow and DrWho17: thanks |
| 14:26.10 | DaPrivateer | ManxPower still here? |
| 14:26.16 | Dublin_drupaller | thanks nick |
| 14:26.29 | Dublin_drupaller | any pointers for more info. on that front? |
| 14:26.34 | Faithful | but then... the asterisk source must be patched for the bristuff to work |
| 14:26.53 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 14:27.11 | nick125_lappy | Dublin_drupaller, well, for actually setting up asterisk, voip-info.org is good |
| 14:27.25 | DrWho17 | Dublin_drupaller: user roled interfaces for asterisk are pretty sparce |
| 14:27.29 | Dublin_drupaller | okay..thanks nick..will check it out....appeciate it. |
| 14:27.37 | Faithful | Why bother... ? because the debian implementtion of asterisk is nice and the packeges are pretty complete |
| 14:27.47 | tzafrir_laptop | Faithful, so I gave you a link to Sarge packages with bristuff patched in |
| 14:28.00 | Nugget | what did debian change? |
| 14:28.02 | tzafrir_laptop | Rebuild them |
| 14:28.18 | *** part/#asterisk Dublin_drupaller (n=dub@83-70-36-157.b-ras1.prp.dublin.eircom.net) |
| 14:28.30 | tzafrir_laptop | Nugget, for once, buildinng debs out of zaptel |
| 14:28.42 | |dennis| | Aother Question: The T1..is an fxo card as well i assume and the only and big difference between the t1 and the regular fxo cards is the on board echo cancellation and hence the huge cost difference? Am I correct? |
| 14:28.45 | tzafrir_laptop | A number of fixes for other architectures |
| 14:28.50 | Nugget | *nod* |
| 14:28.59 | tzafrir_laptop | And some different file locations |
| 14:29.28 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 14:30.08 | tzafrir_laptop | |dennis|, T1 and FXO are totally different |
| 14:30.09 | ManxPower | |dennis|: you are wrong. |
| 14:30.28 | ManxPower | and FXO card supports 1 channel. A T-1 card supports 24 channels. |
| 14:30.42 | DaPrivateer | ManxPower - i tried what you said. the line flashes but when it tries to dial i get a message that hte card is congested. any ideas? |
| 14:30.45 | ManxPower | A T-1 card also support both FXO and FXS (since it's all digital) |
| 14:30.46 | tzafrir_laptop | ManxPower, 24? 23? |
| 14:31.01 | file[laptop] | 23 for a PRI |
| 14:31.03 | |dennis| | ah ok....thank you ppl |
| 14:31.04 | ManxPower | tzafrir_home: 24 channels. On a PRI one of the channels is used for signaling. |
| 14:31.10 | *** join/#asterisk slak- (i=slak@ircdz.com) |
| 14:31.16 | slak- | wa-zup |
| 14:31.17 | DrWho17 | dennis: I carry 24 channels per t1, more efficient, you might even get your 4 lines carried in cheaper |
| 14:31.21 | Faithful | tzafrir_laptop: Thanks that's a winner |
| 14:31.34 | slak- | so yea we're thinking about junking asterisk here ;( |
| 14:31.41 | slak- | because of echo |
| 14:31.52 | slak- | they cant deal with the 5second echo period |
| 14:31.54 | Faithful | I think I should scrub my box and start again now. |
| 14:31.55 | DrWho17 | what does asterisk have to do with echo? |
| 14:32.02 | slak- | zaptel |
| 14:32.51 | |dennis| | talking about echo....if i get the four port fxo TDM04B, use IP phones ......will echo be a problem? |
| 14:33.04 | kyoo | If I purchase a g.729 license from digium - can a remote user with a SPA-2000 or a SNOM phone take advantage of it? (IE, Use it to connect to the asterisk box) |
| 14:33.15 | kyoo | I don't understand where the codec actually *is*... |
| 14:33.18 | DrWho17 | never noticed any echo issues myself, generally it's somewhere else besides asterisk anyway |
| 14:33.23 | DaPrivateer | 9 |
| 14:33.27 | DaPrivateer | oops |
| 14:33.42 | ManxPower | |dennis|: echo is ALWAYS a problem |
| 14:33.54 | drray | either copper wire or lag |
| 14:33.56 | ManxPower | slak-: So fix the echo. |
| 14:34.05 | DrWho17 | I don't have any analog hops |
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| 14:34.07 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 14:34.14 | file[laptop] | darumkilla! |
| 14:34.15 | DrWho17 | tdm/sip or iax |
| 14:34.21 | ManxPower | There are several ways to do it. The easiest is to drop $1,000 on a hardware echo canceler, as someone talked about YESTERDAY on the mailing lists. |
| 14:34.23 | slak- | ManxPower its not working out too well |
| 14:34.31 | drumkilla | hey file[laptop] :) |
| 14:34.37 | ManxPower | DrWho17: So you never call someone on an analog phone? |
| 14:34.37 | |dennis| | ManxPower: Which is best way to deal with it? because if we are to move to asterisk, admin will want to make sure that voice quality and line condition is same or almost equal to what we presently have....so what do you suggest doing? |
| 14:34.52 | slak- | suck a dong |
| 14:34.59 | slak- | forget about phone conversation |
| 14:35.02 | ManxPower | slak-: What hardware echo can are you using? |
| 14:35.17 | ManxPower | |dennis|: the best way to deal with it is a hardware echo canceler. |
| 14:35.18 | slak- | uhm none? |
| 14:35.23 | ManxPower | Not Digium, however. |
| 14:35.27 | slak- | i use the quad fxo module |
| 14:35.29 | slak- | digium |
| 14:35.34 | slak- | and i use software echo cancellers |
| 14:36.37 | |dennis| | slak-: it seems that you have a setup quite like what i want to have.....so how bad is the echo really? |
| 14:37.05 | ManxPower | slak-: Ah. I use echocancel=yes echotraining=yes, lower my txgain until it's as alow as it can go and still have the far end hear us. I also use the ECHO_CAN_MARK3 echo can |
| 14:37.09 | drray | I used 4 FXO ports and got very little echo |
| 14:37.47 | slak- | well |
| 14:37.50 | ManxPower | HOWEVER, we are experimenting with tellabs hardware echo canceler. |
| 14:37.57 | slak- | i just replac ed my * box with a more powerful machine |
| 14:38.00 | slak- | and i think it got better |
| 14:38.08 | slak- | i set my tx gain to 8.0 |
| 14:38.10 | slak- | and it got beter |
| 14:38.11 | slak- | better |
| 14:38.22 | ManxPower | slak-: and if you set your txgain to -8? |
| 14:38.27 | slak- | never tried |
| 14:38.42 | slak- | can i set it and restart * without killing any current phone conversations |
| 14:38.43 | ManxPower | well, the lower your outgoing volume the less echo you'll have. |
| 14:38.52 | ManxPower | slak-: nol |
| 14:39.01 | slak- | echocancel=128 |
| 14:39.01 | slak- | echocancelwhenbridged=yes |
| 14:39.01 | slak- | echotraining=500 |
| 14:39.01 | slak- | relaxdtmf=no |
| 14:39.01 | slak- | rxgain=8.0 |
| 14:39.01 | slak- | txgain=2.5 |
| 14:39.26 | ManxPower | slak-: I already gave you my settings. Don't flood the channel |
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| 14:39.35 | ManxPower | and your txgain is 2.5 not 8 |
| 14:39.36 | slak- | alritey |
| 14:39.42 | slak- | oh i mean rx |
| 14:39.55 | ManxPower | You don't care what the rxgain is on the FXO ports. |
| 14:40.09 | ManxPower | (for echo can issues, that is) |
| 14:40.13 | drray | :) |
| 14:40.26 | slak- | well some echo cancel guide i read on voipinfo suggested that most peopple benefit from around 8.0 |
| 14:40.53 | drray | your txgain is what you need to change |
| 14:40.56 | ManxPower | slak-: 1) the TRANSMITTED audio is what the echo is, so playing with rxgain does nothing for echo. |
| 14:41.03 | *** part/#asterisk bkw_ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 14:41.13 | slak- | ManxPower alrite so you want me to change the tx to -8? |
| 14:41.14 | ManxPower | 2) the value needed for txgain is unique to each location |
| 14:41.18 | slak- | how will that effect the volume |
| 14:41.22 | drray | and each ear |
| 14:41.34 | ManxPower | slak-: you have to play with it while the system is not in production |
| 14:42.00 | slak- | fuck it no ones using it |
| 14:42.10 | mishehu | bah. |
| 14:42.36 | ManxPower | you want to keep lowering your txgain as much as you can but still allowing callers to hear you. |
| 14:42.43 | slak- | haha okay its at -8 |
| 14:43.06 | slak- | crap * crapped out |
| 14:43.10 | ManxPower | then either stop and start asterisk or load/unload chan_zap.so or (CVS-HEAD only) reload chan_zap.so |
| 14:43.15 | slak- | wont even dial with -8 |
| 14:43.29 | ManxPower | slak-: well your volume is prolly too low. |
| 14:43.41 | ManxPower | There is no magic fix, you have to keep experimenting |
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| 14:44.53 | slak- | txgain=-5.0 |
| 14:44.55 | slak- | works okay |
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| 14:45.15 | slak- | no difference interms of echo tho |
| 14:45.29 | ManxPower | slak-: There won't be unless you set everything right. |
| 14:45.50 | slak- | it went away 100% after like 15 seconds |
| 14:45.51 | ManxPower | i.e. changing the echo canceler in zconfig.h, rebuild zaptel, change the other echo cancel options |
| 14:47.43 | ManxPower | I have 9 asterisk servers and don't have echo after applying the settings I gave you earlier. |
| 14:47.52 | ManxPower | slak-: Ah. I use echocancel=yes echotraining=yes, lower my txgain until it's as alow as it can go and still have the far end hear us. I also use the ECHO_CAN_MARK3 echo can |
| 14:48.01 | ManxPower | sorry, echotraining=900 |
| 14:48.09 | *** join/#asterisk blitzrage (n=leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:48.17 | slak- | 900? |
| 14:48.20 | slak- | damn |
| 14:49.16 | slak- | u found that echo_can_mark3 works best? |
| 14:49.26 | ManxPower | slak-: for OUR setups, yes. |
| 14:50.06 | *** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net) |
| 14:50.30 | slak- | can i call u i want to test echo :D |
| 14:50.33 | tzanger | I use MARK2 and optimize zaptel for the proc I'm working on |
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| 14:50.41 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:50.48 | slak- | whats the default one |
| 14:50.51 | tzanger | MARK2 |
| 14:51.25 | coppice | all the echo cancellers in * suck. you just have to find the lesser of several evils |
| 14:51.57 | nroej | all echo cancellation sucks, but this one sucks less *g* |
| 14:52.59 | coppice | not really true. I've done echo cancellers that left everyone deeply impressed. when we did the early ones for *, though, the main goal was low compute rather than good results |
| 14:53.03 | slak- | whats the difference between echotraining 500 and 900 |
| 14:53.07 | slak- | what does it really mean |
| 14:54.31 | drray | so my channel bank would have echo cancelation on it? |
| 14:55.08 | tzanger | drray: I've never heard of htat |
| 14:55.26 | tzanger | coppice: when you die can I have your brain? |
| 14:55.54 | tzanger | slak-: I suggest you do some rudimentary reading, you're asking very basic questions that are very well documented in the wiki and in the doc/ directory of the asterisk source tree |
| 14:56.17 | coppice | drray: channel banks are mostly designed for PSTN connection. they don't need echo can. they need a low price tag |
| 14:57.04 | tzanger | drray: channel banks generally spend their money on good hybrid design which eliminates a lot of their designed echo opportunities |
| 14:57.24 | tzanger | drray: when you throw the T1 into a computer and introduce PCI bus latencies and other overhead you create more opportunities for echo |
| 14:57.37 | coppice | tzanger: no they don't. echo just isn't that important in a channel bank |
| 14:58.47 | tzanger | coppice: I've never had bad echo problems due to poor hybrids in channel banks -- the adit600s I tend to prefer also have marketing mojo which suggests dynamic impedance matching which isn't too far-fetched, although the silabs parts that the TDM modules uses are supposed to do something similar |
| 14:58.54 | coppice | there is only one driver in channel bank design - <$ |
| 14:59.30 | tzanger | coppice: I dunno about that -- there are cheap channel banks and expensive ones, and the featureset runs the gamut between the two ranges |
| 14:59.47 | coppice | the silabs parts don't do dynamic balancing, but they are tunable |
| 15:00.34 | coppice | tzanger: I used to run a team doing line cards. there was only *one* driver, and that was for public exchanges, not some grotty box |
| 15:00.36 | tzanger | that's what I was getting at -- the tunability... perhaps not automatic but being able to tune to me means it's dynamic. :-) |
| 15:00.44 | mishehu | bah. |
| 15:00.47 | bkw_ | I hear lots of complaints about the digium echo can board |
| 15:00.49 | DrWho17 | drray: you probably need a echo cancellation card for your channel bank |
| 15:01.00 | drray | well, I have very little echo |
| 15:01.03 | bkw_ | mishehu, you having fun with OPAL too? |
| 15:01.04 | DrWho17 | ok |
| 15:01.04 | bkw_ | hehe |
| 15:01.13 | mishehu | bkw_: You've Got Mail<tm> |
| 15:01.23 | DrWho17 | carrier access and adtran both have them |
| 15:01.24 | mishehu | heh |
| 15:01.34 | coppice | someone has a tuning program for the tdm400. dunno ho good it is. he asked me how to do it. dunno if he took my advice. |
| 15:01.36 | drray | I just read here and the mailing list and get a creeping fear that one day I will wake up with echo |
| 15:01.51 | mishehu | bkw_: we're having pwlib fun at the moment. I give kudos to craig, it's actually a pretty nice API |
| 15:02.12 | bkw_ | yes even I understood it |
| 15:02.17 | *** join/#asterisk luba (n=chatzill@altom.net1.nerim.net) |
| 15:02.23 | bkw_ | but I don't understand C++ as well as I should |
| 15:02.33 | bkw_ | its like 5 lines of code to setup an IAX2 endpoint |
| 15:02.34 | coppice | bkw_ what do you hear about the digium ec board? |
| 15:02.36 | bkw_ | like 12 for SIP |
| 15:02.57 | bkw_ | coppice, I just see on the lists people are saying they hear clicks and pops and all kinds of weirdness with the hardware echo can |
| 15:02.57 | tzanger | yeah fxotune... that's for tuning the FIR filter |
| 15:03.06 | kyoo | When one person calls into my Asterisk system (calls BV number and gets sent into asterisk via SIP) his keys make no tones - he cannot choose any menu options. His dtmf works on other calls (but he says this does happen to specific numbers) and other are able to use my menus just fine ... Is there anything I can do? |
| 15:03.13 | mishehu | bkw_: I can be of help, of course. c++ is my main favorite language. tony j. (guy who works with me) knows c++ and java also. |
| 15:03.27 | coppice | pwlib seems to have been the biggest problem with openh323. it kept breaking things |
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| 15:03.48 | mishehu | I was thinking of writing a c++ application and mentioning a greets to kram in it, just to raz him a bit... since he so despises c++. |
| 15:04.02 | bkw_ | coppice, but pwlib has been around for almost 20 years hasn't it? |
| 15:04.04 | ManxPower | mishehu: most sane people do. |
| 15:04.17 | coppice | tzanger: i think it changed name. isn't called not the fxotune or something? |
| 15:04.39 | tzanger | coppice: that's what I said. :-) |
| 15:04.49 | mishehu | ManxPower: most sane ppl do what? |
| 15:04.50 | tzanger | or are you saying it's not called fxotune anymore? I don't have any FXO port so I don't really know |
| 15:05.02 | coppice | i said "not the fxotune" :-) |
| 15:05.10 | ManxPower | mishehu: most sane people hate C++ for realtime or OS usage. |
| 15:05.13 | bkw_ | coppice, does pre19 fix anything major? |
| 15:06.29 | DaPrivateer | Grrr... I am trying to preform a PBX transfer with a digium X100P (sip phone). anyone have any suggestions? please? i need to get this done |
| 15:06.29 | tzanger | heh |
| 15:06.32 | tzanger | "Multiply in your head" (ordered the compassionate Dr. Adams) |
| 15:06.32 | tzanger | "365,365,365,365,365,365 by 365,365,365,365,365,365. He [ten-year-old |
| 15:06.32 | tzanger | Truman Henry Safford] flew around the room like a top, pulled his |
| 15:06.32 | tzanger | pantaloons over the tops of his boots, bit his hands, rolled his eyes |
| 15:06.32 | tzanger | in their sockets, sometimes smiling and talking, and then seeming to be |
| 15:06.35 | tzanger | in an agony, until, in not more than one minute, said he, |
| 15:06.38 | tzanger | 133,491,850,208,566,925,016,658,299,941,583,255!" An electronic |
| 15:06.40 | tzanger | computer might do the job a little faster but it wouldn't be as much |
| 15:06.43 | tzanger | fun to watch. |
| 15:06.44 | mishehu | ManxPower: well, I don't even suppose that c++ is an end-all-be-all, it has its place... |
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| 15:06.56 | tzafrir_laptop | regardless of C++, it just seems that pwlib tries to do too many things |
| 15:07.24 | tzafrir_laptop | Somewhat like QT |
| 15:07.27 | mishehu | tzanger: that a dave barry quote? |
| 15:07.35 | tzanger | mishehu: no it's from my fortune file |
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| 15:08.04 | mishehu | tzafrir_laptop: pwlib is more or less just an abstraction layer. do with it what you want to do with it, basically. |
| 15:08.08 | mishehu | ignore the rest. |
| 15:08.29 | bkw_ | its like libc |
| 15:08.32 | bkw_ | just a lib |
| 15:08.51 | mishehu | tzanger: and dave barry is in fortune a lot... the style seemed like him. |
| 15:09.05 | bkw_ | ok now to go buy a book c++ |
| 15:09.17 | tzafrir_laptop | mishehu, look at what recently happened to QT |
| 15:09.43 | mishehu | tzafrir_home: what recently happened to QT ? |
| 15:10.01 | mishehu | all I know is they released a new ver, and have an open source win32 build for it as well... |
| 15:10.23 | tzafrir_laptop | mishehu, and last time I heard, you don't even have to link all of libc in. -lm sounds familiar? |
| 15:11.27 | tzanger | mishehu: I know, but this wasn't him, I forget the attribution |
| 15:12.04 | tzafrir_laptop | mishehu, in QT4 they have finally followed gtk in that they split their library into several sub-libs. So now yo can sanely build console QT apps |
| 15:12.43 | mishehu | tzafrir_laptop: how many sub packages did they split to? |
| 15:13.36 | mishehu | bkw_: ok, so you Don't Have Mail... it bounced on me heh |
| 15:13.49 | tzafrir_laptop | Not sure exactly. I'm more familiar with the gtk side. I know quite a few programs that use glib, pango etc. and not gtk |
| 15:14.25 | *** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca) |
| 15:14.32 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 15:15.02 | coppice | bkw_ if the bug that fixed is one hurting you its always major :-) |
| 15:15.05 | jaike | anyone using polycom 301? were looking for headsets that match those |
| 15:16.02 | coppice | bkw_ pre19 builds cleanly on x86_64 machines. It tolerates FAX machines where the timing is a big off somewhere better |
| 15:16.18 | bkw_ | oh I need to update that |
| 15:16.29 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 15:16.46 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 15:17.26 | tzafrir_laptop | spandsp pre19? |
| 15:18.15 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 15:19.03 | *** part/#asterisk zedkatuf (n=audela@82-46-92-246.cable.ubr08.azte.blueyonder.co.uk) |
| 15:20.28 | ManxPower | I really hope I don't have pneumonia |
| 15:20.35 | slak- | smoke a cig |
| 15:20.38 | slak- | and some ganja |
| 15:20.40 | slak- | best cure |
| 15:20.55 | ManxPower | ganja only treats the symptoms |
| 15:20.56 | coppice | ManxPower: don't worry. you can't pass it on to us |
| 15:21.04 | slak- | man |
| 15:21.04 | *** join/#asterisk JerJer (n=JerJer@pdpc/supporter/bronze/jerjer) |
| 15:21.17 | ManxPower | coppice: but I want to share! |
| 15:21.17 | slak- | i have to setup a 64bit chroot env on this linux box |
| 15:21.21 | slak- | anyone have experience with that |
| 15:22.08 | *** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 15:22.29 | *** join/#asterisk Derkommissar (n=alberto@66.64.215.6.nw.nuvox.net) |
| 15:23.43 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 15:23.46 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@altom.net1.nerim.net) |
| 15:24.54 | *** join/#asterisk pbxbart (n=pbxbart@p54B008D9.dip0.t-ipconnect.de) |
| 15:25.01 | *** part/#asterisk pbxbart (n=pbxbart@p54B008D9.dip0.t-ipconnect.de) |
| 15:25.22 | JerJer | sorry i've only done 8 bit |
| 15:25.29 | tzafrir_laptop | slak-, on debian there is debootstrap for that... |
| 15:26.06 | slak- | yea what do you suggest, install the 64bit port right off the bat or install it ontop of the 32bit |
| 15:26.16 | slak- | like with a chroot |
| 15:26.22 | tzafrir_laptop | slak-, chroot for what? on what distro? |
| 15:26.33 | slak- | debian |
| 15:26.42 | slak- | pure64 |
| 15:27.03 | tzafrir_laptop | slak-, why not ask on #debian ? |
| 15:27.28 | *** join/#asterisk nagl (n=nagl@137.208.4.166) |
| 15:28.02 | *** join/#asterisk Luba_ (n=chatzill@altom.net1.nerim.net) |
| 15:28.02 | olivier_ | and with your nick you shuld install a slackware ;-) |
| 15:28.25 | *** join/#asterisk blake__ (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 15:28.35 | bkw_ | Å“ |
| 15:28.48 | Faithful | tzafrir_laptop: I'm giving that rapid cd a whirl see what I think of it ... if it detects my HFC Bri card I will be impressed |
| 15:29.18 | *** join/#asterisk audela (n=audela@82-46-92-246.cable.ubr08.azte.blueyonder.co.uk) |
| 15:29.44 | Faithful | what is pure64 as opposed to amd64? |
| 15:30.00 | tzafrir_laptop | isn't amd64 pure64? |
| 15:30.07 | Qwell | Faithful: pure64 as opposed to a hybrid 32/64 |
| 15:30.28 | tzafrir_laptop | that is: not s hybrid 64/32 port like the ones of most distros |
| 15:30.32 | *** join/#asterisk newmember (n=newmembe@S010600a0c93dce87.cg.shawcable.net) |
| 15:31.09 | Faithful | Hmmm. so debian-amd64 is not pure? |
| 15:31.24 | Faithful | and when you say hybrid what are you meaning? |
| 15:31.39 | lathos42 | Anyone have an idea why i'm not getting ringback calling out through my SPA-3000 with my Polycom IP501, when I get ringback from my Sipura SPA-841? |
| 15:32.31 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
| 15:32.37 | heison | ~seen sivana |
| 15:32.39 | jbot | sivana is currently on #asterisk (1d 9h 26m 39s) |
| 15:35.29 | lathos42 | Whoops, I'm not getting any audio back at all on my IP 501 |
| 15:37.30 | *** join/#asterisk nroej (n=joern@lak-39-56.wohnheime.ruhr-uni-bochum.de) |
| 15:37.34 | nroej | re |
| 15:38.38 | tzafrir_laptop | Faithful, debian's amd64 port is "pure". That is: it does not contain ia32 compatibility libs |
| 15:38.50 | tzafrir_laptop | there is no /lib and /lib64 |
| 15:39.02 | coppice | that must suck rather badly |
| 15:39.15 | coppice | most people have lib and lib32 |
| 15:39.52 | coppice | FC4 has a mix, though :-\ |
| 15:39.55 | tzafrir_laptop | coppice, maybe. I have no idea. I only played with an amd64 system for a short while about two years ago |
| 15:40.13 | *** join/#asterisk nroej (n=joern@lak-39-56.wohnheime.ruhr-uni-bochum.de) |
| 15:40.14 | nroej | re |
| 15:40.15 | coppice | my X2 machine is very nice, and very fast |
| 15:40.45 | tzanger | X2? |
| 15:40.49 | Qwell | dual core? |
| 15:40.58 | coppice | the dual core AMD64 |
| 15:41.07 | *** join/#asterisk Juxt (n=Juxt@64.135.20.202) |
| 15:41.12 | *** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co) |
| 15:41.21 | tzanger | ah |
| 15:42.04 | *** join/#asterisk gclark (n=gclark@66-193-73-162.gen.twtelecom.net) |
| 15:42.06 | coppice | I have an X2 4200+ machine, and a dual xeon 2.4GHz machine. spandsp runs several times as fast on the X2, but I am not yet clear why it is so much better |
| 15:43.11 | tzafrir_laptop | coppice, a related topic, which I could not get answers to on the mailing list |
| 15:43.31 | shido6 | christ |
| 15:43.34 | shido6 | 4200+ |
| 15:43.35 | tzafrir_laptop | DEbian had a habbit of building everything with 386 compatibility. |
| 15:43.36 | Qwell | yes? |
| 15:43.37 | gclark | Hello all. |
| 15:44.19 | tzafrir_laptop | I was wondering what are the minimal parts I need to rebuild optimized to gain maximum extra performance. |
| 15:44.22 | Juxt | hello |
| 15:44.29 | gclark | I am having a problem with CVS-Head. I installed it but when I try to start asterisk via the init script I receive permission denied. I am running this as root? Any thoughts? |
| 15:44.38 | tzafrir_laptop | I was also wondering what could be a useful benchmark |
| 15:45.16 | tzafrir_laptop | "show translations" is a useful benchmark because it is easily accessible on every asterisk installation. Not sure about the value of the numbers |
| 15:45.21 | Qwell | tzafrir_home: glibc and the kernel would probably give the most noticable boost |
| 15:45.39 | tzafrir_laptop | gclark, permissions denied to where, exactly? |
| 15:45.42 | jaike | is possible to set the recording filename for queue calls to include the extension number that will receive the call? |
| 15:45.52 | tzafrir_laptop | Qwell, what about codecs? |
| 15:46.02 | Qwell | all of them :p |
| 15:46.16 | gclark | I am trying to run '/etc/init.d/asterisk start' |
| 15:47.44 | *** join/#asterisk rkvarala (n=rkvarala@59.93.96.168) |
| 15:47.52 | gclark | The exact error message is 'Starting asterisk: execvp: Permission denied' |
| 15:48.21 | tzafrir_laptop | also: libgsm uses (dynamically) libgsm. how much of the work is done in the asterisk module itself and how much in libgsm? anybody experimented in that? |
| 15:48.38 | tzafrir_laptop | gclark, /usr/sbin/asterisk is not executable? |
| 15:48.51 | *** join/#asterisk aVaLaNcHe (n=stecnic@m85-94-161-25.andorpac.ad) |
| 15:48.59 | Katty | hmm. |
| 15:49.06 | aVaLaNcHe | bzzz |
| 15:49.11 | Katty | i have dumb question - no surprise there |
| 15:49.16 | tzafrir_laptop | 'bash -x /etc/init.d/asterisk start' will give you a trace |
| 15:49.30 | Qwell | Katty: no such thing as a stupid question |
| 15:49.47 | Katty | Qwell: there is :P |
| 15:49.49 | gclark | No. the init script that is created from make config and it is put into /etc/init.d/asterisk and a flag is start|stop|restart |
| 15:49.56 | Qwell | nah, just stupid people :P |
| 15:50.08 | ManxPower | Qwell: I was going to say that. |
| 15:50.09 | Katty | this thing i'm quoting is going to go Broadband -> Firewall -> Router or Switch -> Asterisk -> Phones |
| 15:50.23 | Katty | But i don't know if I need a router or a switch. |
| 15:50.35 | Qwell | does your firewall route? |
| 15:50.35 | Katty | i get that a router..uhh...routes things. |
| 15:50.39 | jaike | gclark: y not just run asterisk -vvvvc? |
| 15:50.41 | tzafrir_laptop | gclark, something is wrong here. Do you want to isolate the problem? |
| 15:50.45 | Katty | but in terms of the firewall routing.... |
| 15:50.48 | Katty | i don't know what i'm looking for |
| 15:50.54 | Katty | so, how do i know if my firewall is also a router? |
| 15:51.13 | *** join/#asterisk charles___ (n=charles@64.35.168.55) |
| 15:51.25 | charles___ | you |
| 15:51.26 | Qwell | I guess if it forwards packets to a switch, it would be a router, right? |
| 15:51.26 | ManxPower | Katty: MOST firewalls also route, especially the cheap ones. |
| 15:51.38 | charles___ | hey guys |
| 15:51.43 | Katty | ManxPower: that doesn't help me determine anything. |
| 15:51.48 | aVaLaNcHe | When I try to make a external call my debug says .. |
| 15:51.51 | Katty | Qwell: i have a model number of a firewall i want to use. |
| 15:51.57 | *** join/#asterisk tAURUS (n=_enver_@213.227.207.45) |
| 15:51.58 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:51.58 | Qwell | oh, you don't have it yet... |
| 15:52.05 | Juxt | has anyone ran asterisk in daemontool? |
| 15:52.11 | charles___ | is there a way to decrease the ammount of rtp on SIP ? |
| 15:52.18 | aVaLaNcHe | quit |
| 15:52.32 | tzafrir_laptop | Juxt, does daemontools work for UDP? |
| 15:52.48 | Qwell | tzafrir_laptop: daemontools is an init replacement thing, afaik |
| 15:53.01 | *** part/#asterisk gr0mit (n=w10277@fw.mot-tools.co.uk) |
| 15:53.19 | *** join/#asterisk aVaLaNcHe (n=stecnic@m85-94-161-25.andorpac.ad) |
| 15:53.24 | aVaLaNcHe | bzzz |
| 15:53.53 | aVaLaNcHe | When I try to make a external call my debug says ... |
| 15:53.54 | aVaLaNcHe | Aug 18 12:24:01 WARNING[3507]: chan_zap.c:7138 zt_request: Unable to determine channel for data {Zap/g1/840064 |
| 15:53.58 | aVaLaNcHe | Aug 18 12:24:01 NOTICE[3507]: app_dial.c:777 dial_exec: Unable to create channel of type 'Zap' |
| 15:54.01 | aVaLaNcHe | Aug 18 12:24:11 WARNING[3507]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' |
| 15:54.04 | aVaLaNcHe | in context 'default' |
| 15:54.06 | aVaLaNcHe | .. |
| 15:54.12 | aVaLaNcHe | anyone can help me ? please |
| 15:54.13 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 15:54.15 | *** join/#asterisk jdg (n=jdg@CA03F8ED.adsl.mana.pf) |
| 15:54.32 | ManxPower | aVaLaNcHe: You have a { where a ( should be |
| 15:54.48 | funxion | good cal ManxPower |
| 15:54.58 | aVaLaNcHe | ManxPower: where ? |
| 15:55.05 | ManxPower | data {Zap/g1 |
| 15:55.07 | funxion | where you dial your zap channel |
| 15:55.20 | ManxPower | you ONLY use { and } for variable names |
| 15:55.27 | gclark | asterisk -vvvvvvvvvc |
| 15:57.03 | punker- | does anybody knows any site where i can find some asterisk with mysql documentation? |
| 15:57.12 | Qwell | punker-: the wiki |
| 15:57.13 | Qwell | ~docs |
| 15:57.13 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 15:57.20 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:57.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:57.33 | ManxPower | ~mailinglist |
| 15:57.33 | jbot | mailinglist is probably Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 15:58.58 | rkvarala | what is the newextension in Agentcallbacklogin |
| 15:59.09 | rkvarala | does any body have any idea on this? |
| 15:59.56 | punker- | thanx |
| 15:59.59 | funxion | newextension? |
| 16:00.26 | *** part/#asterisk jdg (n=jdg@CA03F8ED.adsl.mana.pf) |
| 16:01.17 | aVaLaNcHe | ManxPower: ok, thx that's write now ... but it continue saying Aug 18 12:24:01 WARNING[3507]: chan_zap.c:7138 zt_request: Unable to determine channel for data {Zap/g1/840064 |
| 16:01.20 | aVaLaNcHe | Aug 18 12:24:01 NOTICE[3507]: app_dial.c:777 dial_exec: Unable to create channel of type 'Zap' |
| 16:01.23 | aVaLaNcHe | Aug 18 12:24:11 WARNING[3507]: pbx.c:1952 ast_pbx_run: Timeout, but no rule 't' |
| 16:01.26 | aVaLaNcHe | in context 'default' |
| 16:01.28 | funxion | when you use the agentcallbacklogin you specify the callacbk extension during login |
| 16:01.30 | ManxPower | aVaLaNcHe: no, that's not right. |
| 16:01.31 | aVaLaNcHe | sorry |
| 16:01.34 | ManxPower | you still have a { in there. |
| 16:01.34 | aVaLaNcHe | sorry |
| 16:01.50 | funxion | aVaLaNcHe did you reload? |
| 16:02.05 | aVaLaNcHe | only de line pbx.c 1952 ast_pbx_run Timeout no rule 't' |
| 16:02.21 | ManxPower | well but a hangup after your dial then |
| 16:02.38 | tzafrir_laptop | BTW: isn't there a better way to serch the list archives? e.g: what do you think about gmane? |
| 16:02.46 | ManxPower | ~mailginlist |
| 16:02.49 | ManxPower | ~mailinglist |
| 16:02.49 | jbot | extra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 16:03.10 | tzafrir_laptop | ManxPower, yes, I've just erad that text above |
| 16:03.10 | gclark | tzafrir_laptop, do you have any ideas on the asterisk init script? |
| 16:03.12 | Juxt | does anyone have a list of all area codes and city, state ? |
| 16:03.20 | ManxPower | tzafrir_home: never heard of is. Is it the GNU Lion Hair or something like that? |
| 16:03.35 | ManxPower | Juxt: no. Area codes span miltiple cities and states. |
| 16:03.51 | tzafrir_laptop | gclark, does asterisk start when you run it manually? asterisk -cvvvv |
| 16:03.55 | funxion | juxt you want bellcoe v and h tables |
| 16:04.07 | tzafrir_laptop | (someone already asked you. I probably missed the answer) |
| 16:04.07 | Juxt | ok where do i find htem |
| 16:04.14 | gclark | tzafrir_laptop, yes it runs fine |
| 16:04.22 | funxion | they usually cost money |
| 16:04.24 | ManxPower | Juxt: you purchase them for large amounts of money. |
| 16:04.38 | *** join/#asterisk [kwoot] (n=kvirc@i2rs-son.xs4all.nl) |
| 16:04.47 | citats | heh, its only a few grand for a one time copy :) |
| 16:04.54 | Juxt | heh ok |
| 16:04.57 | tzafrir_laptop | gclark, so with what parameters is it run in the init script? |
| 16:05.23 | DaPrivateer | ManxPower - any chance you can help me a little more with the pbx transfer problem? |
| 16:05.26 | funxion | I remember it being 600 bux last time I updated my list |
| 16:05.34 | tzafrir_laptop | gclark, use script to get the output of that bash -x' line to a file |
| 16:05.36 | Juxt | http://www.telcodata.us/telcodata/downloads |
| 16:05.43 | Juxt | doesn't seem all that pricey |
| 16:05.48 | tzafrir_laptop | and pastebin it |
| 16:05.56 | charles___ | ManxPower: is there a way to decrease the amount of samples per second over SIP ? |
| 16:05.58 | Juxt | lata-npa-nxx-state-city-zip.csv |
| 16:06.12 | charles___ | ManxPower: I can see here that it's overkilling |
| 16:06.17 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 16:06.19 | gclark | start() { |
| 16:06.19 | gclark | <PROTECTED> |
| 16:06.19 | gclark | <PROTECTED> |
| 16:06.19 | gclark | <PROTECTED> |
| 16:06.19 | gclark | <PROTECTED> |
| 16:06.19 | ManxPower | charles___: Why in the world would you ever want to do that? |
| 16:06.19 | gclark | <PROTECTED> |
| 16:06.21 | gclark | <PROTECTED> |
| 16:06.23 | gclark | <PROTECTED> |
| 16:06.25 | gclark | <PROTECTED> |
| 16:06.26 | tzafrir_laptop | ~pb |
| 16:06.26 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 16:06.27 | gclark | <PROTECTED> |
| 16:06.29 | gclark | <PROTECTED> |
| 16:06.31 | gclark | <PROTECTED> |
| 16:06.33 | gclark | <PROTECTED> |
| 16:06.34 | charles___ | ManxPower: because 8 sip channels over G723 is using 1MB |
| 16:06.35 | gclark | <PROTECTED> |
| 16:06.36 | ManxPower | gclark: DON'T FLOOD THE CHANNELS!!!!!!! |
| 16:06.37 | Qwell | ~pastebin |
| 16:06.37 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 16:06.37 | gclark | <PROTECTED> |
| 16:06.38 | Qwell | asshat |
| 16:06.39 | charles___ | 1Mb |
| 16:06.39 | gclark | <PROTECTED> |
| 16:06.40 | citats | nanpa has some text files as well that you can download but they have limited info |
| 16:06.41 | Juxt | gee yesterday i got kicked for that! |
| 16:06.41 | gclark | <PROTECTED> |
| 16:06.43 | gclark | } |
| 16:06.44 | ManxPower | charles___: so switch to a better codec. |
| 16:06.54 | funxion | kewl |
| 16:06.58 | aVaLaNcHe | ManxPower: asterisk -vvvvvc says ... Called g1/800246 , Channel 0/1 span 1 got hangup, Hungup 'Zap/1-1', No one is available to answer at this time |
| 16:07.03 | ManxPower | Juxt: can we jick him again |
| 16:07.04 | *** join/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 16:07.11 | ManxPower | DaPrivateer: no. |
| 16:07.25 | charles___ | ManxPower: man, the same codec over IAX is doing 80Kbps |
| 16:07.40 | ManxPower | aVaLaNcHe: sounds like you are on a PRI and you are not checking the result of the dial |
| 16:07.47 | tzafrir_laptop | gclark, run: script logfile |
| 16:08.01 | ManxPower | charles___: That is correct. |
| 16:08.03 | rkvarala | ManxPower do you have any idea on this Agentcallback login new location |
| 16:08.09 | tzafrir_laptop | gclark, then run bash -x /etc/init.d/asterisk start |
| 16:08.14 | [kwoot] | anyone able to help a newbie? want to connect asterisk to isdn/bri using modem. version is cvs-head from 13 aug 04. |
| 16:08.18 | ManxPower | charles___: and you are not running G723 |
| 16:08.33 | ManxPower | charles___: want to try describing your situation accuratly this time? |
| 16:08.35 | aVaLaNcHe | ManxPower: yes, but I don't know the solution ... can you help me ? |
| 16:08.37 | tzafrir_laptop | run exit. and paste the file logfile in http://pastebin.ca |
| 16:09.03 | tzafrir_laptop | ~docs |
| 16:09.03 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 16:09.16 | ManxPower | aVaLaNcHe: I can do hand holding for $120/hr. |
| 16:09.27 | aVaLaNcHe | thx |
| 16:09.28 | aVaLaNcHe | xDD |
| 16:09.35 | [kwoot] | got to go. |
| 16:09.35 | ManxPower | aVaLaNcHe: otherwise read the wiki about PRI and look at the std-exten in extensions.conf.sample |
| 16:09.43 | aVaLaNcHe | ok, thx |
| 16:09.54 | tzafrir_laptop | [kwoot], read the wiki (woip-info) about modems. which modem exactly? I believe chan_capi and misdn are supported with HEAD |
| 16:10.12 | tzafrir_laptop | bah |
| 16:10.12 | ManxPower | and understand that when you dial out of a PRI you will NEVER hear a busy siglal from the telco, Dial will end the call and set DIALSTATUS = BUSY |
| 16:10.51 | ManxPower | then you have to generate the correct tones to the caller using the dialplan, an example of all this is in std-exten |
| 16:10.52 | bkw_ | ManxPower, that also depends |
| 16:11.06 | ManxPower | bkw_: in 1.0.x I can't get priindication=inband to work |
| 16:11.24 | bkw_ | as I said it depends on how the PRI is provisioned also |
| 16:11.34 | ManxPower | *nod* |
| 16:11.36 | punker- | which is the best linux for asterisk? |
| 16:11.46 | ManxPower | punker-: stop trolling |
| 16:12.00 | nicox | Hello, do anybody know how to cut a dtmf-stream in asterisk? |
| 16:12.01 | ManxPower | punker-: the best linux for ANYTHING is "what you are familiar with" |
| 16:12.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:12.28 | ManxPower | I'm joking. The best Linux is Mandrake/Mandriva linux 9.2 |
| 16:12.30 | bkw_ | ManxPower, stop being a fucking prick |
| 16:12.40 | bkw_ | punker-, use what you're comfortable with |
| 16:12.42 | brad_mssw | whew, just went live today with asterisk ... finally get to trash that POS phone system we did have ... |
| 16:12.49 | punker- | oohhh thanx :) |
| 16:12.51 | *** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 16:13.18 | Ariel_ | wow best linux question |
| 16:13.20 | MikeJ[Laptop] | brad_mssw, what did you have? |
| 16:13.21 | punker- | but i asked that cos some linux are more vulnerables to hackers right?? |
| 16:13.33 | bkw_ | punker-, WRONG |
| 16:13.35 | brad_mssw | MikeJ[Laptop]: talkswitch PBX ... |
| 16:13.39 | Ariel_ | punker-, which one do you like to use? |
| 16:13.39 | tzafrir_laptop | punker-, why is that? |
| 16:13.42 | bkw_ | linux is only as secure as the admin makes it |
| 16:13.47 | punker- | well.. i used fedora |
| 16:13.58 | MikeJ[Laptop] | well, that is probably less secure ;) |
| 16:14.03 | Ariel_ | ok then use fedora or RH or CentOS they all work with Asterisk. |
| 16:14.04 | punker- | i installed asterisk in fedora core 3 |
| 16:14.08 | blitzrage | punker-: if you like RH style - use CentOS, Fedora is just a bitch |
| 16:14.11 | nicox | anybody how can help? |
| 16:14.15 | Qwell | bkw_: unless you use lindows, or whatever the hell they call it now |
| 16:14.19 | Qwell | Lindora? heh |
| 16:14.27 | nicox | Hello, do anybody know how to cut a dtmf-stream in asterisk? |
| 16:14.33 | punker- | but i was reading a documentation of asterisk on devian |
| 16:14.43 | tzafrir_laptop | problem with fedora is that it will not stay supported long enough. Unless fedoralegacy starts improving real-soon-now |
| 16:14.44 | Ariel_ | devian..... |
| 16:15.09 | Ariel_ | if you like Fedora and want something stable that works great use CentOS |
| 16:15.19 | coppice | nicox: tried a carving knife? |
| 16:15.22 | punker- | :O |
| 16:15.22 | tzafrir_laptop | nicox, we're busy in a distro fight. But ask your Q anyway |
| 16:15.38 | punker- | CentOS???? |
| 16:15.53 | blitzrage | punker-: www.centos.org |
| 16:15.55 | Beirdo | tzafrir_home: don't hold your breath about fedoralegacy :) |
| 16:15.56 | punker- | yeah |
| 16:15.59 | coppice | the best linux to use in *not* the one you are familiar with |
| 16:16.00 | nicox | Hello, do anybody know how to cut a dtmf-stream in asterisk? |
| 16:16.03 | punker- | im right ther |
| 16:16.11 | tzafrir_laptop | nicox, can you get the stream in a var and use Cut? |
| 16:16.20 | coppice | its the one some other sucker is familiar with, and you get them, to do all the work |
| 16:16.38 | nicox | i don't know which var the dtmf stream is... thats my problem |
| 16:16.57 | tzafrir_laptop | Beirdo, which is why I don't understand why people base their Asterisk servers (that should remain running for long) on it |
| 16:17.06 | punker- | ok... i have several options... CentOS... Mandrake... Devian.... |
| 16:17.17 | Ariel_ | coppice, nice |
| 16:17.18 | tzafrir_laptop | even worse: some are even installing new asterisk installations on RH9 still |
| 16:17.23 | syle | go fedora core 4 or freebsd |
| 16:17.25 | tzafrir_laptop | Debian! |
| 16:17.29 | nicox | i get a dtmf-stream which comes from the e1 as overlap digits... but with this stream at the end ob the number my telco says nono |
| 16:17.31 | tzafrir_laptop | not Devian |
| 16:17.38 | punker- | ahh ok... no devian |
| 16:17.50 | Beirdo | as I wanted to strip shit down and only compile what I wanted |
| 16:18.07 | nicox | debian is okay, all my asterisks are running on debian *g* |
| 16:18.09 | Beirdo | I use Ubuntu for all my other new installs |
| 16:18.10 | punker- | some friend told me that freebsd is a good linux |
| 16:18.20 | jaike | good linux? |
| 16:18.22 | Ariel_ | argh |
| 16:18.22 | Beirdo | freebsd != linux |
| 16:18.28 | *** join/#asterisk nagl (n=nagl@137.208.4.161) |
| 16:18.28 | Beirdo | freebsd = blech |
| 16:18.28 | Ariel_ | freebsd is not linux |
| 16:18.33 | syle | freebsd > linux |
| 16:18.37 | blitzrage | Hardware won't work on FreeBSD |
| 16:18.38 | Beirdo | they are totally different |
| 16:18.38 | jaike | punker...try winxp |
| 16:18.39 | rkvarala | iam getting this error in Agentcallbacklogin chan_agent.c:1298 __login_exec: Extension '1000' is not valid for automatic login of agent '1000' |
| 16:18.44 | jaike | :P |
| 16:18.49 | syle | hardware works fine on fbsd |
| 16:18.54 | Ariel_ | ROFL winxp |
| 16:18.54 | punker- | FREESB |
| 16:18.55 | blitzrage | Asterisk is developed on, and for Linux. If you're going to use FreeBSD, you better be good :) |
| 16:18.59 | Beirdo | freebsd can eat me |
| 16:19.00 | Beirdo | :) |
| 16:19.01 | Beirdo | hehe |
| 16:19.12 | punker- | Linux Freesb |
| 16:19.38 | *** join/#asterisk Nukemizer (n=Nuke@67.137.28.165) |
| 16:19.56 | Ariel_ | <PROTECTED> |
| 16:19.56 | *** join/#asterisk Blake0PS (n=blake@blakeops.com) |
| 16:20.06 | Qwell | You hit enter instead of r? |
| 16:20.12 | Qwell | Ariel_: You were just a tad off there |
| 16:20.16 | Beirdo | do SIP->IAX :) |
| 16:20.21 | Ariel_ | Qwell, yes |
| 16:20.43 | Ariel_ | Qwell, new ergonomic stupid keyboard. Need my old flat one back. |
| 16:21.16 | Blake0PS | With 2 TDM400Ps installed, which one will be ports 1-4 and which one will be ports 5-8? |
| 16:21.21 | lathos42 | Beirdo: I have * running on my WRT54G at home that i've successfully used to route a call to my * server here at work |
| 16:21.29 | Beirdo | cool |
| 16:21.55 | Beirdo | my use would be to hook up Sipura devices, and to trunk over the internet via IAX |
| 16:22.04 | Beirdo | I don't like SIP particularly |
| 16:22.05 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 16:22.05 | Beirdo | :) |
| 16:22.38 | Beirdo | if I'm careful, I won't need to translate codecs |
| 16:22.48 | nicox | do anybody know if there is a variable where i can find dtmf-digits? |
| 16:22.48 | Beirdo | as I'm sure that will suffer on a wrt |
| 16:22.59 | lathos42 | Beirdo: That's what I did.. I have a SPA-841 that I registered with the WRT, and then used IAX to call between * |
| 16:23.05 | Beirdo | :) |
| 16:23.16 | lathos42 | Beirdo: It worked fine for me using ulaw all the way through the call |
| 16:23.24 | Beirdo | yeah, that was my plan |
| 16:24.18 | nicox | someone who can help? |
| 16:24.32 | jaike | lathos: ur spa841 working fine? we bought around 20 and 5 are defective now |
| 16:24.52 | Beirdo | except I'll be using the SPA3000 |
| 16:25.03 | lathos42 | jaike: Yeah, it seems to be working fine.. I only have one at the moment |
| 16:26.03 | jaike | lathos: were using ip301s right now...they working fine |
| 16:26.48 | nicox | helllo? |
| 16:27.02 | lathos42 | I've got a strange problem at the moment where the IP501 can't hear any audio when I try to call out through the PSTN of the SPA-3000.. but I can get audio just fine if I call any of my phones |
| 16:27.49 | brad_mssw | lathos42: are you going through NAT, I had that issue yesterday |
| 16:28.14 | lathos42 | brad_mssw: Nope, they're both on the same network.. plugged into the same switch on my desk actually |
| 16:28.26 | nicox | do anybody know if there is a variable where i can find dtmf-digits? |
| 16:28.41 | brad_mssw | lathos42: wow, i don't know what the ip501 is, is it also a sip device ? |
| 16:28.46 | brad_mssw | if so, that doesn't make sense |
| 16:28.58 | lathos42 | brad_mssw: Yeah, its a Polycom SIP phone |
| 16:29.33 | *** join/#asterisk J[SS] (i=ph33r@smartserv.ipv6.smart-serv.net) |
| 16:29.47 | DaPrivateer | let me ask a different question. if im in an AGI that just answered a call on a zaptel FXO card, how can i dial a number on that card? |
| 16:30.06 | DaPrivateer | ie just play DTMF tones to the caller |
| 16:30.47 | coppice | how come Dell charge so much less in .us than in asia? damned annoying |
| 16:31.02 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:31.08 | *** join/#asterisk DannyF (n=wizard@c-6d4fe353.24-0099-74657210.cust.bredbandsbolaget.se) |
| 16:31.13 | *** join/#asterisk uma62 (n=sudhir@pool-71-114-93-22.washdc.dsl-w.verizon.net) |
| 16:31.38 | nicox | there is a dtmf stream from the incomming E1 and i will forward the call, but the dtmf-stream is also forwarded, and i don't know hot to stop this stream |
| 16:32.05 | shido6 | whatever you do |
| 16:32.09 | shido6 | dont cross the streams |
| 16:32.31 | coppice | that would be bad, wouldn't it? |
| 16:32.48 | nicox | i have to cut the stream, but i don't know how |
| 16:32.57 | mut | pinch really hard |
| 16:33.05 | Qwell | sharp knife |
| 16:33.09 | coppice | "I am the GPG keymaster" |
| 16:33.11 | coppice | "I am the GNU Gatekeeper" |
| 16:33.24 | citats | lol |
| 16:33.26 | *** join/#asterisk cypromis (n=michael@83.149.70.59) |
| 16:33.30 | shido6 | I am Zuul. |
| 16:33.48 | Hmmhesays | anyone using serweb in here? the install file says nothing about database setup |
| 16:33.57 | nicox | nobody an idea? |
| 16:34.59 | Beirdo | bah humbug |
| 16:35.08 | *** join/#asterisk illvm_ (n=illumina@cpe-65-185-103-95.woh.res.rr.com) |
| 16:35.45 | illvm_ | can anyone help me get a polycom IP500 to connect to my * box? |
| 16:36.32 | blitzrage | illvm_: you should ask specific questions to help you get it setup. People can't hold your hand. |
| 16:36.47 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 16:37.04 | nicox | how can i stop a dtmf-stream? |
| 16:37.42 | coppice | pull the plug out |
| 16:37.44 | illvm_ | Ok... The phone is behind a NAT, I used the web portal to configure the external IP address on the phone and that seems to be working correctly. However, even though the password and username are correct, I still get a 401 message back from asterisk. |
| 16:38.30 | *** join/#asterisk sbnelson (n=sbn@pm3-10stl5.xtraport.net) |
| 16:38.39 | nicox | good joke coppic |
| 16:38.49 | illvm_ | http://pastebin.com/340897 |
| 16:39.30 | |dennis| | which ip phone does one recommend to be best used in a SOHO( a small community school actually) setting.. |
| 16:40.23 | shido6 | PAP2-NA's and analog phones |
| 16:40.31 | shido6 | IAXy's and analog phones |
| 16:41.39 | shido6 | SPA-841 |
| 16:41.50 | sbnelson | Asterisk crashes every so often; is this the place to discuss? |
| 16:41.59 | shido6 | yes |
| 16:41.59 | sbnelson | my asterisk server, I should say. |
| 16:42.15 | |dennis| | shido6: so with the PAP2-Na's i can use my existing phones ..hmm..right? |
| 16:42.23 | shido6 | yes |
| 16:42.23 | sbnelson | It always happens after getting chan_zap.c: Unable to get index, and nullok is not asserted |
| 16:42.25 | shido6 | and it has 2 ports |
| 16:42.31 | shido6 | so 2 users can use it simultaneous |
| 16:42.36 | shido6 | around $70 each |
| 16:42.43 | illvm_ | so does anyone have any ideas why the Polycom SoundPoint IP 501 doesn't register? |
| 16:42.50 | shido6 | yes |
| 16:42.52 | shido6 | check your pm |
| 16:43.01 | shido6 | its your sip.conf |
| 16:43.11 | nicox | is there a chance to change a dtmf stream? |
| 16:43.20 | Derkommissar | from sip.conf , the file |
| 16:43.20 | Derkommissar | how can tell it to open res_odbc ? |
| 16:43.27 | shido6 | and can u get me a screncapture of your config for the polycom or make it accessible from teh pub net |
| 16:43.52 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 16:45.39 | blop | what would u recommand as rackable hardware for asterisk (used with a tdm400p inside of it) ? |
| 16:45.54 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 16:46.08 | *** join/#asterisk luke-jr__ (n=luke-jr@user-0c938q3.cable.mindspring.com) |
| 16:46.15 | |dennis| | shido6: thanks.. |
| 16:46.18 | jaike | illvm: i can dcc you a config doc for polycoms if u like |
| 16:46.22 | blop | sth with the pci port in the front in place of the back would be nice (to patch with the zaps) |
| 16:47.04 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 16:47.55 | sbnelson | Asterisk (CVS-HEAD-04/16/05-14:27:21) dies with chan_zap.c: Unable to get index, and nullok is not asserted -- does anyone have a suggestion? |
| 16:48.17 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 16:48.20 | sbnelson | actually dies sometime afterwards. |
| 16:48.29 | jaike | my only problem with polycoms is that volume resets after each call...looking for a way to make it permanent |
| 16:48.35 | coppice | "Dear Asia-Pacific customer. Dell thinks its bedtime in Asia, and doesn't want to sell you anything" |
| 16:48.56 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 16:49.00 | coppice | jaike: the lack of an echo can in the handset is a common complaint about polycoms |
| 16:49.18 | *** join/#asterisk Gamentine (i=WinNT@d66-222-224-51.abhsia.telus.net) |
| 16:49.49 | jaike | coppice: those are the ones good enough for our budget..ciscos are expensive |
| 16:50.12 | coppice | i wouldn't call polycoms cheap |
| 16:50.48 | nicox | is there a chance to change a dtmf stream? |
| 16:51.36 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:51.55 | *** join/#asterisk secure75 (n=mic@85.233.35.180) |
| 16:52.19 | coppice | jaike: it could be the echo is why they keep reseting the volume |
| 16:52.47 | jaike | To conform to regulatory requirments, |
| 16:52.48 | jaike | handset and headset volume |
| 16:52.48 | jaike | will return to a preset level after each |
| 16:52.48 | jaike | call. Hands-free volume settings will |
| 16:52.48 | jaike | be maintained across calls. |
| 16:53.26 | jaike | crap |
| 16:53.53 | *** join/#asterisk clyrrad (n=ddd@CPE0050bae8d02c-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 16:53.56 | coppice | dunno any regulation which calls for that. it will lessen the impact of echo if you don't make every call loud, though |
| 16:54.20 | mover | can i get the echocancel board thats on TE406P for my TE405P from digium? |
| 16:54.30 | nicox | is there a chance to change a dtmf stream? |
| 16:55.05 | mover | i cant get any infos. support not answered yet. |
| 16:55.35 | drumkilla | mover: yes, but you would have to ship your board back in to have it upgraded |
| 16:55.51 | coppice | jaike: unless the mean the approval for minimum return loss. that would be a nasty twist on true meanings. the sort of thing to make a marketer proud :-) |
| 16:55.59 | mover | drumkilla really? |
| 16:56.04 | drumkilla | yes |
| 16:56.09 | *** join/#asterisk tzanger_ (n=tzanger@mixdown.ca) |
| 16:56.21 | mover | argh... |
| 16:56.33 | coppice | my guess is they need to put the V2 FPGA code in at the same time |
| 16:56.37 | jaike | coppice: theyre using regulations to hide echo problems i guess |
| 16:57.33 | sbnelson | asterisk is dying; only clue is "chan_zap.c: Unable to get index, and nullok is not asserted" sometime before it crashes. Does anyone have an idea on what I can look for? |
| 16:57.34 | mover | coppice ... u are familar with ss7. i got desperate echo on one ss7 machine with libisup. i see this in IAM: [___0____] Octet 1: Echo control device indicator (0=outgoing echo control not included, 1=is included |
| 16:57.49 | mover | is this the cause where echo come? |
| 16:58.00 | mover | on gsm calls now echos happened |
| 16:58.15 | nicox | is there a chance to change a dtmf stream? |
| 16:58.53 | jaike | sbnelson: install asterisk from scratch...thats what i do |
| 16:59.00 | *** join/#asterisk docelm0_ (n=docelm0@67.106.194.90.ptr.us.xo.net) |
| 16:59.16 | sbnelson | jaike: I did! |
| 16:59.24 | coppice | mover: strange the GSM system should have removed the echo before the audio gets to you. what ss7 kit are you using? |
| 16:59.49 | *** join/#asterisk sivana_ (n=sivana@mixdown.ca) |
| 17:00.14 | *** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net) |
| 17:00.15 | SpaceBass | howdy |
| 17:00.18 | mover | libisup TE405P on a Dual Xeon 2.8GHz (32bit SuSE 9.3) |
| 17:00.59 | mover | coppice echo is only on caller side never on called side |
| 17:01.03 | *** part/#asterisk esi (n=ewaldirc@simonis.xs4all.nl) |
| 17:01.25 | *** join/#asterisk _blop (n=blop@213-193-176-119.adsl.easynet.be) |
| 17:01.25 | coppice | which side is the caller? the GSM or the PSTN? |
| 17:01.40 | mover | caller is IP Side |
| 17:02.01 | mover | sip->*->pstn = SIP hear echo |
| 17:02.16 | coppice | so, the caller hears his voice echoing back from the GSM phone? |
| 17:02.22 | mover | sip->*->pstn = SIP hear echo from himself exactly |
| 17:02.52 | mover | coppice no on sip->GSM no echoes on SIP side only on SIP->PSTN |
| 17:03.14 | mover | calls |
| 17:03.17 | mover | :) |
| 17:03.39 | coppice | it makes no sense to turn off the echo can on the GSM system. I didn't think any ISUP selection would do that. maybe i'm wrong |
| 17:04.02 | nicox | HEEEEEEEEEEELLLLLLLLLLPPPPPPPPPP |
| 17:04.16 | mover | coppice u dont understand me |
| 17:04.38 | jontow | question.. |
| 17:04.45 | coppice | mover: you mean the GSM calls are OK. its only SIP<->PSTN that has trouble |
| 17:04.50 | jontow | asterisk seems to be doing ANI, not caller ID on my PRI going to my voicemail system |
| 17:04.52 | *** join/#asterisk cypromis (n=michael@83.149.70.59) |
| 17:04.58 | mover | i have desperate echos from my owhn voice when i call a pstn phone via a snom over libisup |
| 17:05.07 | nicox | is there a chance to change a dtmf stream? |
| 17:05.10 | jontow | which is all fine and well except im gonna have to disable it, i think.. the public service commission will not be dealing with my insolence there |
| 17:05.25 | jontow | a number with blocked caller ID rings in with the correct number when they leave a voicemail |
| 17:05.32 | jontow | .. and apparently that is VeryBad(tm) |
| 17:05.49 | jontow | has anyone had to deal with this? |
| 17:06.00 | sbnelson | nicox: what do you mean? |
| 17:06.13 | coppice | mover: OK. without an echo can you probably will have that problem. |
| 17:06.31 | mover | coppice echocan is on for calls even are bridged |
| 17:07.12 | coppice | mover: it sounds like that is not really the case |
| 17:07.48 | nicox | i have a e1 with incomming calls, and i will forward a call to another E1, but asterisk do not only dial the number which i say, asterisk dial also the exten as dtmf digits after the correct number |
| 17:07.57 | mover | cpatry Echo Cancellation: 128 taps, currently OFF |
| 17:08.20 | nicox | so. exten => 123,1,dial(zap/g1/456) is dialed as 456123 |
| 17:08.31 | *** part/#asterisk tAURUS (n=_enver_@213.227.207.45) |
| 17:08.33 | nicox | and 123 are diald as dtmf digits |
| 17:08.53 | mover | coppice what idea you have???? |
| 17:09.21 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:09.31 | coppice | you said ec is on, and the said currently off. which is it? |
| 17:09.58 | mover | currenty off is because channel is ofhook |
| 17:10.10 | mover | its onhook echocanel is on |
| 17:10.13 | sbnelson | nicox: what are the exact lines from extensions.conf? I do the same thing on a T1 and don't have ur troubles. |
| 17:10.20 | mover | wait i paste an onhook channel |
| 17:10.57 | nicox | do you have overlapdial=yes in zapata.conf? |
| 17:11.02 | mover | Echo Cancellation: 128 taps, currently ON |
| 17:11.40 | coppice | it sounds like something is not right in the EC config. maybe when the message says is not what is really happening |
| 17:11.48 | nicox | because i need it to get most of the calls, because our POTS send the digits not in one block |
| 17:12.41 | mover | wait i tell you ec conf |
| 17:12.52 | sbnelson | nicox: no |
| 17:13.27 | sbnelson | nicox: I don't think I can help you then. |
| 17:13.33 | *** join/#asterisk cypromis (n=michael@83.149.70.59) |
| 17:13.41 | mover | echocanel=yes echocanelwhenbr.=yes echotraining=yes thats all |
| 17:14.11 | mover | but i have tried training=800 and 600 all the same |
| 17:14.59 | *** part/#asterisk sbnelson (n=sbn@pm3-10stl5.xtraport.net) |
| 17:15.06 | mover | only mark3 and agreessive supression kill the f.. echo. but calls are totally unnatural and to clean an filtered |
| 17:15.37 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 17:16.02 | coppice | ah, so there is some way to kill the echo. at least that means it is being applied. now to figure out why it is normally ineffective |
| 17:16.24 | mover | coppice i dunno |
| 17:16.30 | mover | all the logs are the same |
| 17:16.59 | mover | all the calls are to the same pstn phone and all tries are echoed |
| 17:17.39 | *** join/#asterisk jsmith (n=jsmith@64.50.35.114.2O7.net) |
| 17:17.43 | coppice | what happens if you disable EC? is it the same, better, worse? |
| 17:17.54 | mover | same |
| 17:18.11 | mover | but echo a little clearer |
| 17:18.37 | mover | btw: i use all straight 1-0-9 libpri,zaptel and * |
| 17:18.44 | Katty | hewwo. |
| 17:18.47 | coppice | ah, better quality echo :-) |
| 17:19.05 | mover | coppice exact |
| 17:19.37 | mover | i have played with rx,txgain no success all the same |
| 17:19.52 | *** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it) |
| 17:20.02 | nicox | has anyone an idea to my problem? |
| 17:20.08 | charles___ | Hey, do you guys know any reason to my asterisk be forking in 10 process on start ? |
| 17:20.08 | mover | so what went wrong... §%$§$%@@!### |
| 17:20.38 | Qwell | charles___: 10 asterisk processes? |
| 17:20.47 | charles___ | Qwell: yes, on start |
| 17:20.55 | charles___ | Qwell: my other asterisk is solitaire |
| 17:21.04 | anthm | some linux show threads as processes |
| 17:21.25 | charles___ | Qwell: but this new gets 15 children |
| 17:21.33 | tzafrir_laptop | the latest from Michael Robertson: http://www.michaelrobertson.com/archive.php?minute_id=186 |
| 17:21.50 | *** join/#asterisk w0w0 (n=w0w0@125.Red-83-46-188.pooles.rima-tde.net) |
| 17:21.57 | mover | coppice u thinking or clueless? |
| 17:21.59 | charles___ | anthm: no way man, if it has a PID it's a proccess! |
| 17:22.06 | tzafrir_laptop | So does an * server that should serve a whole campus and includes a T1 card costs 600$? |
| 17:22.45 | Qwell | kram: evening |
| 17:22.52 | tzafrir_laptop | if they have exactly the same memory sizes: threads of the same process |
| 17:22.58 | kram | hi qwell |
| 17:23.05 | Darwin35 | Katty did it I saw here . she used wire cutters and went right to it in the network closet |
| 17:23.15 | Katty | ... |
| 17:23.16 | coppice | mover: dunno. kind of strange. there should be some substantial effect when EC is on |
| 17:23.34 | Katty | kram: feeling better today? |
| 17:24.33 | Darwin35 | man if this softclient copiles and works we will have a sip softphone worth using |
| 17:24.43 | Darwin35 | compiles even |
| 17:24.53 | anthm | charles___ if you are the expert why are you asking, I can assure you that gentoo for instance has it set so threads have pids and show up as processes but since you seem to know better I gues I have no idea. |
| 17:25.37 | mover | coppice change of entire box havent solve the echoes |
| 17:25.53 | kram | uhm, i don't think anything here has changed since yesterday. i have a friend visiting this weekend so i'll just try to spend some time away for a bit |
| 17:26.06 | Katty | kram: good plan (= |
| 17:26.07 | mover | on another box with a TE110P singlespan the same echoes |
| 17:26.18 | coppice | mover: that was pretty unlikely to be a cure |
| 17:27.15 | *** join/#asterisk _PiGreco_ (n=a@adsl-ull-87-69.47-151.net24.it) |
| 17:27.18 | _PiGreco_ | hello |
| 17:27.43 | coppice | threads are like a loving family - they share the same address space :-) |
| 17:27.47 | _PiGreco_ | is it possible to REGISTER to a server different from the one in sip:user@server register line in sip.conf ? |
| 17:28.09 | mishehu | anthm: howdy. |
| 17:28.17 | anthm | allo |
| 17:28.46 | _PiGreco_ | i mean i am user@server but i want to REGISTER to server2 instead, same username |
| 17:28.54 | _PiGreco_ | (i have a kind of dns problem) |
| 17:28.58 | mover | coppice i am despaired ... |
| 17:29.00 | mountie | anthm: Re threads/pids: Linux 2.4 each thread has a PID - in 2.6, they've separated threads/PIDs. |
| 17:29.32 | mountie | anthm: but of course, some distros (like RH,SUSE) have backported parts of 2.6 to 2.4 (making threads share PID) |
| 17:29.43 | mover | i cant reach markku |
| 17:30.09 | coppice | mover: markku might be on vacation |
| 17:30.16 | charles___ | anthm: it is pid's man |
| 17:30.19 | mover | coppice i know |
| 17:30.22 | *** part/#asterisk jluk (n=jluk@82-32-208-99.cable.ubr07.newt.blueyonder.co.uk) |
| 17:30.52 | anthm | yah, your asterisk is rebellious and forks magicly |
| 17:30.59 | charles___ | anthm: root@pc01:/etc/asterisk# ps -ef |grep asterisk |wc |
| 17:30.59 | *** join/#asterisk pbxbart_ (n=pbxbart@p54B02FFE.dip0.t-ipconnect.de) |
| 17:30.59 | charles___ | <PROTECTED> |
| 17:31.10 | bkw_ | mountie, thats Linux threads vs NPTL |
| 17:31.14 | mover | so nevertheless i need to fix it asap |
| 17:31.21 | coppice | anthm speak with forked tongue |
| 17:31.25 | *** part/#asterisk pbxbart_ (n=pbxbart@p54B02FFE.dip0.t-ipconnect.de) |
| 17:31.27 | charles___ | anthm: my asterisk is like a bunny, 1 minute and 20 children |
| 17:31.35 | anthm | ps -ef |grep asterisk |wc |
| 17:31.35 | anthm | <PROTECTED> |
| 17:31.38 | anthm | me too |
| 17:31.51 | mountie | bkw_: Right. NPTL doesn't work on 2.4 (exept on some suse/rh kernels with the kernel support backported) |
| 17:31.51 | anthm | cos i have a gentoo box that shows threads as pids |
| 17:31.57 | tzafrir_laptop | anthm, is that 2.4 or 2.6? |
| 17:32.26 | charles___ | anthm: I have 2.6 here |
| 17:32.27 | *** join/#asterisk [hC] (n=hardcore@8.10.2.5) |
| 17:32.28 | anthm | and on redhat9 there is 1 |
| 17:32.32 | tzafrir_laptop | on 2.6 there is /protc/<PID>/task with the threads data . And ps indeed shows just one entry for asterisk |
| 17:32.34 | charles___ | anthm: 2.6.12 |
| 17:32.38 | anthm | it depends on the distro |
| 17:32.48 | coppice | 2.4 or 2.6 make no difference. its the display software that either groups them or doesn't |
| 17:32.53 | tzafrir_laptop | RH backported it from 2.6 |
| 17:32.55 | bkw_ | root@shinzon [Fri Aug 19 12:32 PM] ~/opal/samples/simple |
| 17:32.55 | bkw_ | <8>:ps aux | grep asterisk | grep -v safe_asterisk | grep -v grep | wc -l |
| 17:32.55 | bkw_ | 1 |
| 17:32.56 | anthm | and the THREADS_HAVE_PIDS or some flag |
| 17:32.59 | mover | coppice so what u were do in my situationÃ? |
| 17:33.03 | Qwell | ps -ef | grep asterisk | wc |
| 17:33.03 | Qwell | <PROTECTED> |
| 17:33.10 | tzafrir_laptop | But then again, 2.6 is becoming more and more common |
| 17:33.11 | [hC] | hey guys.. Any suggestions for what i can do (other than nat=yes and qualify=yes) to keep sip phones that are behind nats from going unreachable every now and then? |
| 17:33.22 | bkw_ | linux threads suck |
| 17:33.30 | bkw_ | nptl is 400% faster |
| 17:33.52 | bkw_ | but can be harder to debug |
| 17:34.07 | *** join/#asterisk Tili (i=Tili@218.19.67.91) |
| 17:34.13 | charles___ | which os uses NPTL ? freebsd ? |
| 17:34.25 | Qwell | linux 2.6 (and backport 2.4?) |
| 17:34.29 | bkw_ | WRONG |
| 17:34.30 | tzafrir_laptop | charles___, linux 2.6 |
| 17:34.51 | Qwell | bkw_: wrong which? |
| 17:34.54 | bkw_ | its glibc that needs to be compiled with NPTL |
| 17:34.56 | bkw_ | not really the kerenl |
| 17:34.58 | bkw_ | er kernel |
| 17:35.09 | Qwell | dunno, thats how gentoo makes it seem |
| 17:35.20 | bkw_ | export USE=nptl |
| 17:35.22 | bkw_ | emerge glibc |
| 17:35.29 | Qwell | export? |
| 17:35.35 | Qwell | USE="nptl" emerge glibc |
| 17:35.36 | Qwell | boi |
| 17:35.39 | bkw_ | same thing |
| 17:35.40 | tzafrir_laptop | bkw_, but it needs kernel-level support as well |
| 17:35.46 | Qwell | easier :p |
| 17:35.48 | anthm | there is no way for asterisk to accidently fork so you can be positive you are seeing threads that is the jist |
| 17:35.53 | tzafrir_laptop | As I have just showed, /proc looks different |
| 17:36.16 | anthm | there are only like 3 places that fork in the whole dist of asterisk |
| 17:36.18 | bkw_ | tzafrir_laptop, I don't think you enable anythign special in the kernel ata ll |
| 17:36.20 | bkw_ | for NPTL |
| 17:36.26 | anthm | 1 to go in the bg for the console |
| 17:36.35 | anthm | 1 in the mpg123 style moh |
| 17:36.44 | tzafrir_laptop | bkw_, does it work with a vanilla 2.4? 2.2? |
| 17:36.52 | bkw_ | 2.6 kernels yes |
| 17:36.56 | bkw_ | lower kernels NO |
| 17:37.02 | tzafrir_laptop | 2.6 already have it |
| 17:37.02 | anthm | and a few more here and there in some apps |
| 17:37.14 | bkw_ | I don't even consider NPTL for 2.4 and lower |
| 17:37.27 | anthm | agi of corse |
| 17:37.41 | *** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 17:37.47 | Derkommissar | bkw_, im having a problem with res_odbc i dont know if its known.... afther a couple of hours when i reload... it doesnt slurp the config.. |
| 17:37.59 | bkw_ | Derkommissar, what country are you in? |
| 17:38.11 | Derkommissar | USA at this momen :-) |
| 17:38.19 | bkw_ | but where are you from normally |
| 17:38.31 | Derkommissar | Im Cuban, but i work in ecuador |
| 17:38.38 | bkw_ | I talked to you on the phone |
| 17:38.40 | nicox | is there anyone who have a good heart? |
| 17:38.45 | Derkommissar | Yes you did |
| 17:38.50 | Derkommissar | im peters friend |
| 17:38.56 | Derkommissar | we talked before though |
| 17:39.00 | bkw_ | yes I put two and two together.. i'm quick like that ;) |
| 17:39.00 | Qwell | nicox: I will when I get my transplant |
| 17:39.03 | Derkommissar | you consulted for me before too |
| 17:39.10 | Derkommissar | :-) |
| 17:39.12 | Derkommissar | LOl |
| 17:39.35 | nicox | please help me... i need help with overlapdial and dtmf digits |
| 17:39.38 | bkw_ | Derkommissar, ok someone has been dinking with the config stuffs |
| 17:39.59 | bkw_ | let me look |
| 17:40.02 | Derkommissar | :-/ what do you mean |
| 17:40.10 | Derkommissar | i cant duplicate it always |
| 17:40.12 | Derkommissar | but sure |
| 17:40.26 | bkw_ | let me look |
| 17:40.28 | bkw_ | hrm |
| 17:40.34 | *** join/#asterisk sigmounte (n=sigmount@85.201.48.109) |
| 17:41.00 | Qwell | ugh, don't msg me |
| 17:41.04 | *** part/#asterisk veteran (n=veteran@cpe-66-25-30-132.houston.res.rr.com) |
| 17:41.15 | shido6 | ? |
| 17:41.21 | coppice | i can't always duplicate, but I so love to practice ;-) |
| 17:42.11 | charles___ | anthm: thanks man, I will look at ntpl |
| 17:42.12 | Qwell | coppice: I think you're thinking of another word. :p |
| 17:42.15 | nicox | <PROTECTED> |
| 17:42.59 | Qwell | coppice: propagate maybe |
| 17:42.59 | Derkommissar | nicox, i like falco, but i dont speak the lenguage |
| 17:43.01 | coppice | Qwell: dunno. the successful duplication efforts produces reasonable replicas |
| 17:43.12 | mover | Derkommissar hehehe |
| 17:45.24 | Beirdo | how about Vienna Calling? |
| 17:45.36 | charles___ | Do you guys know if I NOLOAD all the unnecessary modules will make any performance effect ? |
| 17:45.38 | Beirdo | Falco++ |
| 17:46.53 | blitzrage | charles___: it could a bit. I prefer to use the autoload=no option and specifically load the modules I need. I made a file to do it - its on the wiki at http://www.voip-info.org/wiki-Asterisk+Slimming |
| 17:47.54 | charles___ | blitzrage: thanks |
| 17:48.02 | mover | welche asterisk version ist das? |
| 17:48.16 | *** part/#asterisk Juxt (n=Juxt@64.135.20.202) |
| 17:49.17 | kyoo | I have Asterisk running through a sipura spa-2000 ... can I put people on hold, transfer, etc using an analog phone through this adapter? |
| 17:49.24 | shido6 | yes |
| 17:49.25 | shido6 | you can |
| 17:49.30 | Qwell | call parking, and # transfer |
| 17:49.31 | mover | hmm ich würede mal versuchen 123,1,trnsfer(234) |
| 17:50.05 | kyoo | shido6: Qwell: Where would I find docs for that? |
| 17:50.10 | Qwell | wiki |
| 17:50.16 | shido6 | wiki |
| 17:50.31 | nicox | transfer, is vielleicht ne idee, mahc ich gleich mal |
| 17:50.36 | shido6 | or or hs1wkwtftad |
| 17:50.43 | Qwell | shido6: eh? |
| 17:50.50 | bjohnson | smartass |
| 17:51.02 | bjohnson | heineken |
| 17:51.03 | shido6 | hire some 1 who knows what the fsck they are doing |
| 17:51.14 | shido6 | wyw |
| 17:51.16 | bjohnson | strudel |
| 17:51.18 | shido6 | while you watch :) |
| 17:51.22 | bjohnson | fraulein |
| 17:51.28 | bjohnson | combine and enjoy |
| 17:51.29 | Qwell | al (and learn) |
| 17:51.36 | shido6 | wsc |
| 17:51.39 | shido6 | with screen capture |
| 17:51.44 | charles___ | is there a way to have multiple bindaddr ? |
| 17:51.56 | shido6 | 0.0.0.0 for all |
| 17:52.13 | shido6 | if u have 1 out of 9 devces u dont wanna lsiten on |
| 17:52.19 | shido6 | tis ok to listen |
| 17:52.36 | Qwell | just do 0.0.0.0 and setup iptables |
| 17:52.40 | charles___ | shido6: wanted to listen on 2 only |
| 17:52.40 | shido6 | yep |
| 17:52.42 | *** join/#asterisk psycodad (n=obiwan@2001:4060:4419:b1:0:0:0:2) |
| 17:52.51 | shido6 | what Qwell said |
| 17:53.00 | charles___ | shido6: is 1 or all the possibilities? |
| 17:53.04 | Qwell | --dport 5060 --d 192.168... |
| 17:53.18 | charles___ | Qwell: yes |
| 17:53.26 | charles___ | -j DROPP |
| 17:54.41 | charles___ | bindaddr=IP, IP; doesn't work with space also doesn't. I'm going to assume 1 or ALL |
| 17:54.44 | charles___ | is it? |
| 17:55.05 | *** join/#asterisk coppice_ (n=chatzill@226.193.17.210.dyn.pacific.net.hk) |
| 17:57.11 | Hmmhesays | file oh file where art thou |
| 17:57.34 | file[laptop] | uh oh |
| 17:57.50 | *** join/#asterisk rking (n=rking@ip68-105-231-56.lu.dl.cox.net) |
| 17:57.51 | file[laptop] | what do you want? :P |
| 17:58.20 | rking | heh, me? |
| 17:58.31 | Hmmhesays | being an SER n00b myself, can you tell me if there is a way for ser to tell you why the config file is bad? i'm just getting ERROR: bond config file (2 errors) |
| 17:58.35 | Hmmhesays | *bad even |
| 17:58.42 | file[laptop] | oh, yes ;) |
| 17:58.51 | file[laptop] | ser -c |
| 17:58.51 | blitzrage | asteriskdocs.org totally restored! |
| 17:59.13 | Hmmhesays | wonderful, thanks file[laptop] |
| 17:59.17 | file[laptop] | yw |
| 17:59.46 | blitzrage | http://ask.slashdot.org/askslashdot/05/08/19/1428233.shtml?tid=215&tid=222&tid=4 |
| 18:00.15 | rking | i have asterisk running, and ohphone handy, and i'd like to get either in a state where i can present them to my company as an alternative to skype, but as far as i know i'd need another person on the other end. actually, is there an equivalent to skype's echo123 service? a bot running H323 that will echo sounds back? |
| 18:01.06 | bjohnson | do you have h323 compiled for asterisk? |
| 18:01.19 | bjohnson | that will likely be more of a problem |
| 18:01.44 | bjohnson | there are examples in the demo of playing back a file, time and date, etc |
| 18:02.09 | bjohnson | why use h323 if you don't have to .. use a iax or sip softphone instead |
| 18:02.14 | rking | bjohnson: hrm, i was told "asterisk is voip, with pots if you have the hardware" |
| 18:02.26 | bjohnson | yes that is true |
| 18:02.46 | bjohnson | but the h323 channel driver isn't included with it, just the sip and iax ones |
| 18:02.53 | rking | bjohnson: i completely apologize for my total ignorance of h323 vs iax vs sip. i'm in a greenfield area of exploration on all this. prior to 3 months ago i didn't even own a headset. =) |
| 18:02.56 | bjohnson | you have to compile h323 separately |
| 18:03.08 | bjohnson | it has lots of dependancies and is difficult to compile correctly |
| 18:03.15 | rking | ok, cool |
| 18:03.23 | rking | so linphone? |
| 18:03.37 | bjohnson | easier to use sip or iax unless you already have a h323 system you have to tie intop |
| 18:03.40 | *** join/#asterisk coldfeet (n=cold@dsl-80-46-109-145.access.as9105.com) |
| 18:03.43 | bjohnson | linphone is a sip one |
| 18:03.51 | bjohnson | iaxcomm is a iax one |
| 18:03.54 | bjohnson | there are other |
| 18:03.57 | bjohnson | there are others too |
| 18:04.32 | rking | bjohnson: we do have to support windows and mac users, so maybe sip is the choice? |
| 18:04.51 | Qwell | iaxcomm is cross platform |
| 18:05.16 | tzafrir_laptop | rking, in fact, there are about 4 separate h323 channels right now |
| 18:05.37 | tzafrir_laptop | sip is a protocol, just like IAX and h323. |
| 18:05.48 | tzafrir_laptop | And just as "cross-platform" |
| 18:06.25 | rking | tzafrir_laptop: right... but my [very cursory] search for info to compare iax vs sip seems that sip is more established, and i'm assuming would have better windows/mac support (?) |
| 18:07.00 | tzafrir_laptop | rking, are you sure you want a soft phone? |
| 18:07.33 | rking | tzafrir: we use skype all the time, and it funamentally works great, but skype crashes a lot, and has a 5-person conference limit, and is not open |
| 18:07.43 | tzafrir_laptop | rking, anyway, there are nice phones for both protocols for windows and for mac "as well" |
| 18:08.05 | Ethon | rking: You can plug an asterisk between your trunk and your normal pbx to handle voip with your normal phones |
| 18:08.06 | rking | tzafrir: so you recommend iax? |
| 18:08.55 | tzafrir_laptop | rking, iax would do a much better job if you want to traverse NAT. |
| 18:08.57 | rking | Ethon: yeah, that's totally a goal for the future stuff - but right now we'd be happy to have 100% voip going on. we use freeconference.com for when people have to be POTS'd in, currently |
| 18:09.12 | tzafrir_laptop | If it's inside the same LAN there isn't that much a difference |
| 18:09.21 | rking | tzafrir: awesome, yes - that would be a constant problem. i was figuring siproxy would be in the mix |
| 18:10.05 | Ethon | rking: Was just a solution we made for many of our customers which did not replace their old telephone hardware and use voip |
| 18:10.17 | charles___ | Cpu(s): 67.9% us, 3.6% sy, 0.0% ni, 24.6% id, 0.0% wa, 0.0% hi, 3.9% si |
| 18:10.28 | charles___ | hehe 20 channels over G723 on a 2.4Ghz |
| 18:11.37 | rking | i am so grateful for your help. i do have one area where i am unclear, but feel free to ignore me at this point: |
| 18:12.37 | heison | ~seen sivana |
| 18:12.40 | jbot | sivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 20h 3m 7s ago, saying: 'ya'. |
| 18:12.45 | rking | i'd like to test iaxcomm in isolation for call-out, then i'd like to use iaxcomm+asterisk to make a conference, and then i'd like to show it to someone i work with. is there anyone here i can call out to for just ~30 seconds, and then i'll start fiddling with the asterisk side on my own after that. |
| 18:13.26 | rking | bjohnson mentioned the demo's for playing back files, but i'd first like to eliminate iaxclient variables if possible |
| 18:13.42 | charles___ | Qwell: hey man do you really have 720 threads ? |
| 18:13.46 | Qwell | charles___: no |
| 18:13.51 | charles___ | Qwell: hehehe |
| 18:14.10 | charles___ | Qwell: the machine here is at 70% cpu and it's pretty stable |
| 18:14.59 | rking | is there an online iaxbot somewhere? |
| 18:15.09 | Qwell | rking: to do what? |
| 18:15.10 | Ethon | rking: Sorry, I'm at home.. my internet connection is not fast enough for a demonstration |
| 18:15.15 | charles___ | maybe with NPTL I can compress 1 T1 |
| 18:15.25 | *** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) |
| 18:15.48 | rking | Qwell: so i can use iaxclient to call it and verify that everything is working |
| 18:15.59 | Qwell | rking: guest@24.50.66.194/s |
| 18:16.25 | Qwell | it'll be lagged, but sound is sound |
| 18:17.04 | rking | right... now, if i could figure out iaxcomm... 1 minute |
| 18:17.18 | Qwell | rking: in the dial box, just put what I said |
| 18:18.07 | rking | awesome! |
| 18:18.12 | rking | music to my ears. |
| 18:19.25 | drumkilla | wtf is an iaxbot :) |
| 18:19.32 | Qwell | ~iaxbot |
| 18:19.46 | rking | drumkilla: guest@24.50.66.194/s is definitely what i'd call an iaxbot. |
| 18:19.53 | drumkilla | ha |
| 18:20.44 | rking | now to config asterisk to host a conference call. |
| 18:20.49 | drumkilla | I have heard Qwell's shinanigan's before |
| 18:21.11 | kyoo | What would I search for specifically in the wiki to learn about call control from an adapted analog phone? |
| 18:21.53 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:23.30 | PupenoL | Hello. |
| 18:24.11 | bjohnson | kyoo: you have a supira? search for sipura |
| 18:26.06 | jarrod | hey anyone know of an agi or the likes that can accept a key sequence and set a flag that would notify the extension to forward a call instead of ring the device? |
| 18:26.24 | *** join/#asterisk mag_um (i=cvx@219.95.202.108) |
| 18:26.37 | shido6 | can I dd over a partition without formatting it? |
| 18:26.58 | Qwell | shido6: if you dd from one partition to another, a format isn't needed. it copies the FS |
| 18:27.15 | Qwell | same with disk to disk, but replace fs with partition table (and fs..) |
| 18:27.37 | Qwell | or, do you mean, format it before dd? |
| 18:28.52 | *** join/#asterisk zoo (i=nobody@ip-54-16.travedsl.de) |
| 18:29.31 | brad_mssw | i've got 2 TDM400Ps, one with 4 FXO's one with 4 FXS's ... everything seems to mostly work fine, but it seems that sometimes the phone system misreads the extension numbers entered (especially if someone calls from a cell phone) ... for instance, someone typing 114, might be transferred to 111 instead ... I've verified via the asterisk logs that it says it got 111 ... are there any tweaking settings for these? |
| 18:29.49 | JerJer | shido6: dd is simply a low level copy |
| 18:30.01 | JerJer | it doesn't care, just just does what you tell it to do |
| 18:30.07 | brad_mssw | i've also verified 114 was actually entered as per the history in the cell phone |
| 18:30.50 | ManxPower | brad_mssw: set relaxdtmf=no and play around with your txgain and txgain for the FXO ports. too loud or too soft would cause these problems |
| 18:31.05 | ManxPower | ..er..txgain and rxgain |
| 18:31.07 | brad_mssw | ManxPower: thanks, I'll try it |
| 18:31.26 | greg_work | brad_mssw: ztmonitor can help you set gain |
| 18:31.31 | *** join/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 18:31.45 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 18:31.56 | netnameus | Hi... my box is hanging during bootup on "loading zaptel hardware modules"... how can I get past this point? |
| 18:33.35 | *** join/#asterisk brookshire (n=matt@c-67-190-190-32.hsd1.co.comcast.net) |
| 18:35.02 | bkw_ | netnameus, boot up and remove the zaptel init |
| 18:35.13 | bkw_ | could be any number of things |
| 18:36.29 | netnameus | how do i bootup and remove the zaptel init? |
| 18:36.35 | netnameus | (i'm new to linux) |
| 18:36.42 | bkw_ | take the card out |
| 18:36.51 | netnameus | oh, i dont have a car |
| 18:36.52 | netnameus | card |
| 18:36.59 | brad_mssw | ManxPower: is relaxdtmf default to on or off ? |
| 18:37.01 | bkw_ | then at the boot prompt do linux single |
| 18:37.32 | netnameus | i don't have a zaptel card |
| 18:38.03 | ManxPower | brad_mssw: no idea. |
| 18:39.45 | *** join/#asterisk r0d3nt (n=RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 18:39.52 | brad_mssw | ManxPower: default appears to be 'no' ... perhaps I should turn it on instead ? |
| 18:40.42 | JerJer | you won't know until you try it |
| 18:40.45 | JerJer | test |
| 18:40.48 | JerJer | test again |
| 18:40.55 | JerJer | test more |
| 18:41.09 | brad_mssw | yeah, that's the hard part ... it only happens sometimes :/ |
| 18:41.11 | ManxPower | brad_mssw: no, don't do that |
| 18:41.18 | ManxPower | brad_mssw: you have a volume problem then |
| 18:41.28 | brad_mssw | ok, I'll bark up that tree then, thanks |
| 18:44.18 | Katty | hmm. |
| 18:44.59 | *** join/#asterisk aminorex (n=tony@12-23-137-226.dhcp.dlth.mn.charter.com) |
| 18:45.58 | netnameus | how can i stop it from trying to load the "zaptel hardware moduels"? |
| 18:46.13 | Qwell | netnameus: <bkw_> then at the boot prompt do linux single |
| 18:46.39 | bkw_ | Qwell, thanks for taking care of my light work |
| 18:46.42 | blitzrage | anyone know what a .snp file is? |
| 18:46.49 | Qwell | bkw_: I'll bill you later |
| 18:46.53 | bkw_ | thanks |
| 18:46.58 | bkw_ | ;) |
| 18:47.08 | jarrod | anyone know of a good archive of useful agi scripts? |
| 18:47.18 | Katty | anyone want to sell me 60 phones? |
| 18:47.32 | netnameus | "Error 27: Unrecognized command" |
| 18:47.34 | netnameus | is what i get |
| 18:47.42 | blitzrage | Katty: sure, do I have to actually send you phones, or can I just take the money and run? :) |
| 18:47.53 | Katty | blitzrage: ;P |
| 18:48.13 | jaike | where do u guys buy ur equip? the only site i know is voipsupply.com |
| 18:48.16 | zoa | i will sell you 60 freeware phones :) |
| 18:48.19 | zoa | hey ho all |
| 18:48.31 | Qwell | jaike: there is also that was...umm...starts with an |
| 18:48.31 | Qwell | x |
| 18:48.36 | blitzrage | Katty: :D |
| 18:48.43 | puowvip | Katty, Western Electric model 500? |
| 18:48.53 | *** join/#asterisk Defraz (n=t0tal@24-119-12-238.cpe.cableone.net) |
| 18:48.57 | Katty | 60 phones, cheaper than 54 bucks a piece. |
| 18:49.01 | Katty | those are the stipulations. |
| 18:49.03 | netnameus | i get "Error 27: Unrecognized command" |
| 18:49.07 | Qwell | Katty: any type of phone? |
| 18:49.09 | Katty | and voIP, obviously |
| 18:49.14 | Qwell | I've got these awesome fisher price phones... |
| 18:49.17 | Qwell | wireless |
| 18:49.18 | Katty | Qwell: hardware, not software (= |
| 18:49.29 | Katty | ;> |
| 18:49.36 | zoa | working phones ? |
| 18:49.37 | ManxPower | Katty: if those are the requirements then tell them that you cannot provide a system at that price and find another vendor. |
| 18:49.45 | Katty | ManxPower: shoo (= |
| 18:49.49 | ManxPower | You don't want to work on a project where they are so cheap. |
| 18:49.52 | Katty | ManxPower: you obviously don't see my sarcasm. |
| 18:50.02 | zoa | i dont think you can do it for that price |
| 18:50.02 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 18:50.05 | ManxPower | Katty: I'm 1/2 out of it today. Sick. |
| 18:50.08 | jaike | we might be buying couple of hundred soon...still have to decide between polycoms and snoms |
| 18:50.19 | zoa | snom snom snom |
| 18:50.23 | Katty | snom :< |
| 18:50.25 | Katty | polycom :> |
| 18:50.25 | ManxPower | polycom polycom polycom |
| 18:50.27 | jarrod | i like polycoms |
| 18:50.29 | Qwell | cisco |
| 18:50.30 | puowvip | Western Electric model 500. |
| 18:50.35 | Ariel_ | polycom polycom |
| 18:50.40 | Qwell | jaike: http://www.voip-info.org/wiki-VOIP+sites Use at your own risk |
| 18:50.45 | jaike | polycom 5, snoms 1 |
| 18:50.57 | Katty | we have 13 polycoms and 1 snom |
| 18:51.08 | brad_mssw | what should I dial while using ztmonitor to test the gain, etc? |
| 18:51.21 | MikeJ[Laptop] | ALASKA! |
| 18:51.22 | zoa | oh whatever you do, stay away from thomson |
| 18:51.27 | ManxPower | we have 60 polycoms |
| 18:51.39 | jaike | yah..were leaning on getting polycoms |
| 18:51.41 | zoa | we were interested ins several thousand thomsons, and i ask for latest firmware |
| 18:51.44 | Katty | MikeJ[Laptop]: no, mexico. |
| 18:51.53 | zoa | and the reply is: you cannot do software upgrades on that thing |
| 18:51.54 | zoa | WTF ? |
| 18:52.03 | zoa | how come they have UK firmware in the uk |
| 18:52.06 | Qwell | jaike: ahh, voxilla, thats the one I was thinking of |
| 18:52.07 | ManxPower | zoa: they are not owned by Digium, are you? |
| 18:52.10 | zoa | and they have a web interface for that |
| 18:52.11 | ManxPower | ..., are they? |
| 18:52.17 | zoa | nopez |
| 18:52.36 | jaike | cool..thanks |
| 18:52.50 | zoa | i gave a negative advise for that thing, based oa on that reply |
| 18:53.08 | *** join/#asterisk jorgito (n=jorgito@snat2.arachne.czfree.net) |
| 18:53.09 | jorgito | hi |
| 18:53.40 | *** part/#asterisk netnameus (n=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net) |
| 18:53.48 | *** join/#asterisk xADDY (n=xAD_nFL@host144-199.pool8290.interbusiness.it) |
| 18:53.49 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 18:53.58 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:53.58 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 18:55.27 | *** join/#asterisk Grumbly (n=Mantacul@209.151.244.129) |
| 18:55.43 | Grumbly | hello. I was wondering if I could use asterisk as a replacement for panalog |
| 18:55.49 | Grumbly | if anyone knows what that is |
| 18:55.55 | JerJer | define panalog |
| 18:56.25 | Grumbly | Panalog is a Call Management system / pbx logger. Works with panasonic pbx stuff |
| 18:56.40 | JerJer | like a serial port thing? |
| 18:56.40 | jaike | lathos: thought u were having problems with ur ip 501 :P |
| 18:56.47 | Grumbly | um... lemme look... |
| 18:56.58 | lathos42 | jaike: Its a love hate relationship that I have with it |
| 18:57.18 | Grumbly | indeed... like a serial port thing |
| 18:57.32 | Grumbly | heh... I never looked at the box it was running on... |
| 18:58.04 | Grumbly | it's got 2 serial port devices. |
| 18:58.19 | JerJer | so what do you want to replace? |
| 18:58.23 | JerJer | or perhaps how? |
| 18:58.55 | zoa | you could do it |
| 18:58.56 | Grumbly | I'd Ideally love to replace this P.O.S. software they call a call managment suite... |
| 18:59.00 | zoa | but it will take time |
| 18:59.07 | zoa | but its not asterisk that will do it |
| 18:59.15 | zoa | you need perl |
| 18:59.16 | zoa | thats all |
| 18:59.28 | Grumbly | We're not a voip system though |
| 18:59.40 | JerJer | asterisk can do analog telephones |
| 18:59.45 | zoa | but for the cost of making such a thing, you might as well replace everything with asterisk |
| 18:59.51 | JerJer | it all depends on exactly what you are looking to replace |
| 19:00.08 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 19:00.15 | zoa | o unless its a 100000 phones setup, i'd replace it with asterisk |
| 19:00.22 | zoa | *so |
| 19:00.23 | jaike | anyone using POE on the polycoms? nice feature |
| 19:00.35 | ManxPower | yes |
| 19:00.47 | zoa | asterisk on slashdot again |
| 19:00.51 | zoa | its almost every week now |
| 19:00.55 | jaike | POE switches seem expensive though |
| 19:00.56 | Grumbly | Well JerJer, I'd like to replace the system that logs the inbound and outbound call times, numbers, etc. with a reliable one. It also should be able to automatically route calls based upon time of day |
| 19:01.16 | zoa | you need only perl |
| 19:01.17 | jorgito | listen |
| 19:01.19 | zoa | no asterisk |
| 19:01.20 | jorgito | listen |
| 19:01.32 | jorgito | did anybody tried to compile asterisk for some embeded system.? |
| 19:01.40 | zoa | i did something like that before, its not fun to program |
| 19:01.41 | jorgito | Grumbly, ha ha ha |
| 19:01.55 | jorgito | zoa, this was to me? |
| 19:01.58 | Qwell | Why would you want asterisk on an embedded system? |
| 19:01.59 | zoa | jorgito, listen, listen |
| 19:02.01 | zoa | listen |
| 19:02.02 | zoa | listen |
| 19:02.03 | zoa | listen |
| 19:02.10 | MikeJ[Laptop] | Grumbly, you are talkking about a serial port logger, then you are wanting one that controls calls.. |
| 19:02.14 | MikeJ[Laptop] | we are all missing somthing |
| 19:02.29 | zoa | he wants a 2 way communication over serial port |
| 19:02.47 | jorgito | zoa, i tried to compile it against uclibc and not not working..... |
| 19:02.48 | Grumbly | Mike9: the suite we've got logs calls and controlls the routing |
| 19:02.49 | zoa | brrr |
| 19:02.57 | *** join/#asterisk greenok (n=greenr21@port0006-abm-adsl.cwjamaica.com) |
| 19:02.58 | lathos42 | jaike: We're going to go with PoE when we roll out in production |
| 19:02.58 | zoa | jorgito, i know people did it before |
| 19:03.00 | jorgito | zoa, what library exactly does asterisk needs? |
| 19:03.01 | Grumbly | based on time of day... though I may be mistaken about that |
| 19:03.07 | zoa | jorgito: dunno |
| 19:03.08 | jorgito | zoa, tell me one pleas. . |
| 19:03.09 | Grumbly | It might be a term into the pbx |
| 19:03.16 | lathos42 | jaike: If I can get this problem with the spa-3000 fixed that is =) |
| 19:03.16 | Grumbly | lemme ask |
| 19:03.26 | zoa | google for astlinux |
| 19:03.46 | jorgito | zoa to me? |
| 19:04.01 | zoa | yes |
| 19:04.25 | *** part/#asterisk greenok (n=greenr21@port0006-abm-adsl.cwjamaica.com) |
| 19:04.27 | lathos42 | Has anyone successfully called out through the PSTN with an SPA-3000, calling from a Polycom phone? |
| 19:04.43 | zoa | how do you do that ? |
| 19:04.48 | zoa | lathos ? |
| 19:04.58 | zoa | polycom to spa3000 to pstn ??? |
| 19:05.09 | jorgito | what is spa-3000 ? |
| 19:05.10 | Qwell | spa3000 is an fxo, no? |
| 19:05.18 | zoa | ah it might have fxo yes |
| 19:05.29 | zoa | and fxs |
| 19:05.36 | lathos42 | Yeah, its both |
| 19:05.37 | zoa | but how do you connect a polycom to that ? |
| 19:05.42 | zoa | polycom is voip |
| 19:05.51 | Qwell | zoa: so is the spa3000 |
| 19:05.55 | zoa | yes i know |
| 19:05.59 | *** join/#asterisk darkskiez (n=darkskie@host86-133-151-115.range86-133.btcentralplus.com) |
| 19:06.01 | Qwell | through asterisk, of course |
| 19:06.05 | *** join/#asterisk sigmounte (n=sigmount@85.201.48.109) |
| 19:06.12 | zoa | although it might be possible |
| 19:06.12 | lathos42 | SPA3000 --SIP--> * <--SIP-- Polycom |
| 19:06.16 | zoa | aaah |
| 19:06.19 | zoa | that has no pstn |
| 19:06.19 | Qwell | see? :p |
| 19:06.24 | zoa | aaaaah |
| 19:06.28 | Qwell | its more like |
| 19:06.35 | zoa | you mean phone - fxs on spa3000 |
| 19:06.38 | zoa | yes i get it now |
| 19:06.40 | brad_mssw | what's the best way to set the gains using ztmonitor ? |
| 19:06.42 | Qwell | polycom(sip) > * > spa(fxo-pstn) |
| 19:06.42 | zoa | that should work just fine |
| 19:06.53 | Qwell | polycom(sip) > * > (sip)spa(fxo-pstn) |
| 19:06.54 | Qwell | ratehr |
| 19:06.56 | brad_mssw | and how do you know when it's right, or close to right? |
| 19:07.04 | zoa | brad_mssw: trial and error |
| 19:07.07 | zoa | listen to it |
| 19:07.20 | brad_mssw | zoa: from what i've read, it says not to trust your ears |
| 19:07.28 | Qwell | well... |
| 19:07.29 | brad_mssw | zoa: supposed to do some quantitative thing |
| 19:07.34 | zoa | its hard |
| 19:07.36 | brad_mssw | docs aren't that great though |
| 19:07.37 | Qwell | if there is echo, and it thinks its done |
| 19:07.41 | zoa | as one phone might be loader than something else |
| 19:07.43 | Qwell | I'm gonna say "uhh...no. There is still echo" |
| 19:07.44 | zoa | louder |
| 19:07.52 | brad_mssw | Qwell: echo is not the issue |
| 19:07.59 | Qwell | s/echo/volume/ then |
| 19:08.04 | brad_mssw | Qwell: it's misreading the tones |
| 19:08.10 | brad_mssw | Qwell: sometimes .... |
| 19:08.14 | zoa | aaah |
| 19:08.17 | lathos42 | The crazy thing is that I can call out just fine when I use my Sipura SPA-841, or the SPA-3000's fxs port.. but for some reason when I call with the Polycom it can send but not receive audio.. but if I put the call on hold, it suddenly starts working when the call comes off hold |
| 19:08.21 | brad_mssw | like it'll mistake 114 for 111 |
| 19:08.24 | zoa | are you using the dtmf relax ? |
| 19:08.29 | brad_mssw | no |
| 19:08.30 | jorgito | so nobody here compiled the asterisk to embeded system? |
| 19:08.41 | zoa | jj has it on wrt |
| 19:08.43 | Qwell | jorgito: What purpose would it serve, exactly? |
| 19:09.10 | zoa | lets all pm jj |
| 19:09.40 | darkskiez | so, i'm behind a nat, and I can sign on with a softphone+stun to sipgate, but not with asterisk. It connects and I can make and receive calls, but the RTP doesnt come back. How should I proceed? Can I use SER somehow to do some weird STUN thing? |
| 19:10.04 | *** join/#asterisk Grumbly (n=Mantacul@209.151.244.129) |
| 19:10.07 | Grumbly | eek |
| 19:10.20 | lathos42 | I've got * running on my WRT54G at home |
| 19:10.51 | lathos42 | It makes a nice 1 phone SIP to IAX gateway |
| 19:11.27 | Qwell | no transcoding, heh |
| 19:11.32 | Qwell | and it can't route at the same time |
| 19:11.36 | Qwell | but hey, it works, right? |
| 19:11.45 | *** join/#asterisk tessier (n=treed@se0-0.ar1.tma1.loudpacket.net) |
| 19:11.58 | jorgito | Qwell, well asterisk for embeded system.. exactly i am working in ISP and i would like to port asterisk to our routers, running with uclibc library |
| 19:12.12 | jorgito | Qwell, so crosscompiling the asterisk.. |
| 19:12.36 | tessier | jorgito: Easier to port your routing to a standard asterisk platform. |
| 19:13.15 | jorgito | tessier, maybe |
| 19:13.22 | tzafrir_laptop | jorgito, on what CPU? |
| 19:13.42 | jorgito | tzafrir_laptop, normal i386 family |
| 19:13.54 | tzafrir_laptop | jorgito, if you want to use voip over the internet, you will need a decent CPU |
| 19:14.15 | Qwell | combining routing and asterisk on one machine is kinda silly |
| 19:14.26 | lathos42 | Qwell: I didnt try to transcode, but I was still able to access the internet from my iBook using WPA just fine :) |
| 19:14.27 | blitzrage | hey, I'm running to the store for some food, you guys want anything? |
| 19:14.27 | jorgito | tzafrir_laptop, realy? i dont thing so |
| 19:14.37 | jorgito | Qwell, why not? |
| 19:14.43 | Qwell | lathos42: I was being facetious |
| 19:14.50 | ManxPower | blitzrage: How about a bottle of high strength antibiotic? |
| 19:15.00 | *** join/#asterisk secure75 (n=mic@gfwlan.cablesurf.de) |
| 19:15.03 | Qwell | jorgito: because it shouldn't work that way |
| 19:15.22 | Qwell | jorgito: routers route, PBXs ..PBX |
| 19:15.26 | lathos42 | Qwell: I'd be interested to see which had a higher CPU load during the call.. my WRT54g, or the Polycom phone |
| 19:15.33 | Qwell | Yes, I DID just use PBX as a verb. |
| 19:15.37 | zoa | wrt |
| 19:15.37 | jorgito | Qwell, i dont see a reasont why it should do that... |
| 19:15.40 | zoa | probably |
| 19:15.48 | jorgito | hmmm |
| 19:15.51 | jorgito | i will try.. |
| 19:15.57 | Qwell | ManxPower: mind explaining this better? |
| 19:16.09 | ManxPower | Qwell: not today |
| 19:16.33 | Qwell | jorgito: if I may use an analogy.. |
| 19:16.54 | lathos42 | Qwell: You should submit PBX as a verb to one of the Open Dictionaries on the web |
| 19:17.09 | Qwell | jorgito: Do you have any TVs that toast bread? How about a microwave that can download the latest sport scores? |
| 19:17.17 | mishehu | bah. |
| 19:17.26 | jorgito | Qwell, pssst |
| 19:18.01 | Qwell | or maybe a cellphone that can make you a 7 course meal |
| 19:18.25 | Qwell | a car that also does dishes? |
| 19:18.35 | Qwell | a washing machine that is also a web server? |
| 19:18.43 | Qwell | I can come up with these all day |
| 19:18.48 | RaYmAn-Bx | that would rock though |
| 19:18.53 | jorgito | Qwell, hmmm . you blind? |
| 19:18.55 | Qwell | RaYmAn-Bx: it might, but its useless |
| 19:19.05 | Qwell | jorgito: routing and calling are as different as day and orange |
| 19:19.14 | Qwell | s |
| 19:19.36 | jorgito | Qwell, well linux machine with asterisk must route also ... |
| 19:19.38 | jorgito | anyway |
| 19:19.56 | Qwell | let me guess, it'll also have a GUI? |
| 19:20.08 | heison | ~seen sivana |
| 19:20.09 | jbot | sivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 21h 10m 36s ago, saying: 'ya'. |
| 19:21.07 | jorgito | Qwell, or the asterisk box doesnt route? |
| 19:21.12 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
| 19:21.24 | Qwell | jorgito: nope, it doesn't need to |
| 19:22.21 | jorgito | Qwell, what ? |
| 19:22.26 | tzafrir_laptop | jorgito, what prevents a linus box from routing? |
| 19:22.35 | Qwell | Why would the asterisk box need to be a router also? |
| 19:22.43 | tzafrir_laptop | jorgito, did you try building asterisk with uclibc? |
| 19:22.53 | jorgito | tzafrir_laptop, yes |
| 19:22.58 | tzafrir_laptop | and? |
| 19:23.08 | jorgito | tzafrir_laptop, did you see Makefile ? horrible |
| 19:23.29 | tzafrir_laptop | I'm willing to help thee |
| 19:23.51 | tessier | Anyone remember the single-fxs card that Digium used to sell with their developer kit? That is not a TDM400P card right? I think they had something different before. But still used the little daughterboards. |
| 19:23.51 | jorgito | realy? |
| 19:24.29 | blitzrage | tessier: nope - daughter boards only on TDM400P - previous card was a modem (X101P) |
| 19:24.30 | tzafrir_laptop | jorgito, I want to sanitize it |
| 19:24.35 | blitzrage | tessier: and in the IAXy. |
| 19:24.40 | jorgito | tzafrir_laptop, what is sanitize? |
| 19:24.49 | tessier | blitzrage: hmm.... |
| 19:24.51 | tzafrir_laptop | make it saner |
| 19:25.20 | tzafrir_laptop | jorgito, but do you have a specific error? |
| 19:25.43 | jorgito | tzafrir_laptop, wait i will modify Makefile and then i will try to compile.. |
| 19:25.47 | tzafrir_laptop | did you try building asterisk in a uclibc build chroot? |
| 19:26.04 | tzafrir_laptop | why do you need to modify the makefile? |
| 19:26.26 | jorgito | tzafrir_laptop, to change includes and cc and pthreads etc... |
| 19:27.42 | tzafrir_laptop | jorgito, why not build it in a uclibc chroot? |
| 19:27.44 | *** join/#asterisk Cybertank (n=cybertan@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com) |
| 19:28.20 | jorgito | tzafrir_laptop, you dont need chroot.. |
| 19:28.25 | *** join/#asterisk RoyK (n=roy@ti211210a080-0158.bb.online.no) |
| 19:28.40 | jorgito | tzafrir_laptop, just changing the cc, includes, librarys is enaugh |
| 19:28.50 | tzafrir_laptop | jorgito, but it will make things simpler. It means no cross-compilation settings |
| 19:29.15 | tzafrir_laptop | Also: all the libs you build against are built with glibc, right? |
| 19:29.27 | jorgito | yes , but you know why make it simple if you can make it hard.. |
| 19:30.03 | *** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 19:30.13 | jorgito | tzafrir_laptop, no , the libc are compiled against uclibc |
| 19:30.16 | *** part/#asterisk jorgito (n=jorgito@snat2.arachne.czfree.net) |
| 19:30.22 | *** join/#asterisk jorge_bla (n=jorge@snat2.arachne.czfree.net) |
| 19:30.24 | jorge_bla | hi |
| 19:30.34 | jorge_bla | it s me jorgito |
| 19:30.44 | jorge_bla | /home/LRP/buildroot/build_i386/staging_dir/bin/i386-linux-uclibc-gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libe |
| 19:30.44 | jorge_bla | dit.a db1-ast/libdb1.a stdtime/libtime.a -L/home/LRP/buildroot/build_i386/staging_dir/lib -ldl -lpthread -lncurses -lm -lresolv -lssl |
| 19:30.44 | jorge_bla | dns.o(.text+0x37d): In function `ast_search_dns': |
| 19:30.44 | jorge_bla | /home/LRP/asterisk-1.0.9/dns.c:174: undefined reference to `__res_ninit' |
| 19:30.46 | jorge_bla | dns.o(.text+0x3b2):/home/LRP/asterisk-1.0.9/dns.c:175: undefined reference to `__res_nsearch' |
| 19:30.48 | jorge_bla | dns.o(.text+0x499):/home/LRP/asterisk-1.0.9/dns.c:194: undefined reference to `__res_nclose' |
| 19:30.50 | jorge_bla | collect2: ld returned 1 exit status |
| 19:30.51 | tzafrir_laptop | ~pb |
| 19:30.51 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 19:30.52 | jorge_bla | make: *** [asterisk] Error 1 |
| 19:31.02 | jorge_bla | sorry |
| 19:31.49 | jorge_bla | seems like i didnt defined linker.. |
| 19:32.35 | jorge_bla | http://pastebin.ca/20445 |
| 19:33.18 | mover | where i can se a list what free codecs will supported by asterisk (possibly with a download site) |
| 19:33.37 | bkw_ | mover, the ones listed when you do show translations |
| 19:33.43 | *** join/#asterisk yogurt2ungue (n=yogurt2u@44-170-114-200.fibertel.com.ar) |
| 19:33.45 | bkw_ | with the exception of g723.1 |
| 19:33.48 | jorge_bla | tzafrir_laptop, do you see? |
| 19:33.59 | *** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net) |
| 19:34.54 | tzafrir_laptop | jorge_bla, are openssl and ncurses linked with uclibc? Though this is not the error here |
| 19:35.07 | jorge_bla | yes |
| 19:35.14 | mover | bkw_ thanks |
| 19:35.30 | jorge_bla | root@lrp:/home/LRP/buildroot/build_i386/staging_dir/lib# ls | grep resol |
| 19:35.30 | jorge_bla | libresolv-0.9.27.so |
| 19:35.36 | clyrrad | Hi, can someone please clairfy for me when I put configuration changes into sip.conf and iax.conf? From the samples I have found online they seem to contain the same sorts of items, and a few posts said you can make your configurations in either of the 2 files. Was wondering if someone can please clarify that for me. |
| 19:35.58 | Nugget | http://lnk.nu/westpress.co.uk/3lq.jsp - cessna flies for two hours with half a wing missing |
| 19:36.13 | Nugget | "But despite two of the three passengers being top flight engineers on their way to fix a Boeing 767, no one noticed that half the left wing, containing one fuel tank, was missing." |
| 19:36.17 | Nugget | (with pic) |
| 19:36.44 | *** join/#asterisk iCEBrkr_ (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 19:37.04 | *** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 19:37.12 | mover | anyone here get ibeam from xten work with h26x through *? |
| 19:37.26 | mover | is this possible? |
| 19:37.42 | bkw_ | mover, the best way to know is try it |
| 19:37.53 | bkw_ | there are only a few ways things work or don't work |
| 19:37.59 | mover | i cant i dont have ibeam :-( |
| 19:38.01 | bkw_ | just start trying stuff till it works or doesn't work |
| 19:38.01 | Ariel_ | Nugget, seems that he needs some glasses |
| 19:38.23 | bkw_ | mover, thats how I solve problems.. just jump in and try shit till I get debug that proves it won't or I get it working |
| 19:38.47 | Nugget | I have eyebeam, but I've never tried h26x. |
| 19:38.50 | mover | if i had it i still have tried |
| 19:38.58 | tzafrir_laptop | jorge_bla, can't think of anything solid right now |
| 19:39.21 | jorge_bla | tzafrir_home, me either |
| 19:39.38 | jorge_bla | does anybody know somebody who compiled asterisk against uclibc |
| 19:39.57 | mover | i think its the proto that Ciscos Vidphone use or i am wrong? |
| 19:40.17 | *** join/#asterisk pbd (n=pbdavids@12.144.118.36) |
| 19:41.32 | *** join/#asterisk Lan16`spdy^gTp (n=spdy@lns-th2-15-poi-82-64-230-87.adsl.proxad.net) |
| 19:42.06 | jorge_bla | tzafrir_laptop, seems that lots of people have the saim problem http://lists.digium.com/pipermail/asterisk-dev/2004-December/007831.html |
| 19:42.28 | *** part/#asterisk Lan16`spdy^gTp (n=spdy@lns-th2-15-poi-82-64-230-87.adsl.proxad.net) |
| 19:42.41 | bkw_ | jorge_bla, thats old |
| 19:42.50 | bkw_ | coming up on a year almost |
| 19:43.30 | jorge_bla | bkw_, but still the problem... |
| 19:43.33 | docelm0_ | Anyone know what voltage the TDM400 cards are? |
| 19:43.36 | docelm0_ | 3.3v? |
| 19:43.52 | jorge_bla | i think lots of people doesnt compile asterisk against uclibc |
| 19:44.10 | darkskiez | docelm: dual |
| 19:44.22 | docelm0_ | shweet.. :) thanks! |
| 19:44.32 | Grumbly | ok... |
| 19:44.55 | Grumbly | can asterisk be used solely to LOG call data? |
| 19:45.05 | Grumbly | from a serial device attached to a pbx? |
| 19:45.10 | jsmith | Grumbly: Sure, if you set it up that way... |
| 19:45.19 | *** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net) |
| 19:45.20 | Grumbly | OK |
| 19:45.21 | bkw_ | Grumbly, um search for cdr_serial.c |
| 19:45.22 | Grumbly | sweet |
| 19:45.42 | bkw_ | http://www.dynx.net/ASTERISK/diff-patches/cdr_serial.c |
| 19:45.44 | bkw_ | I wrote that |
| 19:45.45 | bkw_ | as a test |
| 19:45.46 | jsmith | Or just pass the calls from one port to another, and log along the way... |
| 19:45.52 | Grumbly | can asterisk be used to manage a pbx? |
| 19:45.59 | bkw_ | asterisk is a PBX |
| 19:46.02 | Grumbly | i kno |
| 19:46.18 | Grumbly | but we've got a pbx already, we just need a log reader |
| 19:46.28 | Nugget | asterisk is not what you want, then. |
| 19:46.38 | jsmith | Grumbly: As a log *reader*? No.... |
| 19:46.56 | Grumbly | not reader... sorry... I'm doin several things and having several conversations |
| 19:46.57 | Grumbly | heh |
| 19:47.10 | jsmith | Grumbly: As a lot writer, then yes... |
| 19:47.13 | jsmith | s/lot/log/ |
| 19:47.20 | *** join/#asterisk Defraz (n=t0tal@tim.ibccom.net) |
| 19:47.42 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 19:48.45 | Grumbly | I'm looking to replace a call data analyser with the specific function of using serial ports to a pbxswitch to retrieve and write log files of inbound and outbound calls from any extention on the system. |
| 19:48.50 | Grumbly | can * do that |
| 19:49.27 | Grumbly | ? |
| 19:49.44 | drumkilla | that would have nothing to do with asterisk ... |
| 19:49.48 | Nugget | asterisk is a pbx, not an activity monitor. |
| 19:49.48 | Qwell | just write a simple app, no need for a full PBX to do that |
| 19:50.07 | Grumbly | Qwell: I am not so much good with the coding. |
| 19:50.59 | Qwell | You'll need to code with anything you use |
| 19:51.20 | Grumbly | I mean app coding |
| 19:51.22 | Grumbly | not good |
| 19:51.24 | Qwell | You'll need to code with anything you use |
| 19:51.28 | Grumbly | scripting= ok |
| 19:51.31 | rabelais | in sip.conf, in my register line, if my username has an @ symbol in it, how do I let asterisk know about that so it doesn't confuse the symbol with the host of the sip registrar? |
| 19:51.40 | *** join/#asterisk Cybertank_ (n=cybertan@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com) |
| 19:52.02 | Hmmhesays | better than drippy genitals |
| 19:52.53 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:52.56 | Grumbly | bbiam |
| 19:53.44 | nick125_lappy | hmm |
| 19:53.59 | nick125_lappy | i wonder if theres a problem with my asterisk or xlite phone |
| 19:54.21 | nick125_lappy | when i go to 7777, and press 1 (which goes in a queue), the hold music plays for a moment, then stops |
| 19:55.12 | jaike | nick: whats ur moh player? mpg123? |
| 19:55.16 | nick125_lappy | yeah |
| 19:55.22 | Ariel_ | nick125, what does the cli says. |
| 19:55.32 | jaike | make sure its the right one for ur version of asterisk |
| 19:55.41 | nick125_lappy | Ariel_, hold on |
| 19:55.49 | Qwell | jaike: version of asterisk doesn't matter. 0.59r period |
| 19:56.05 | Ariel_ | nick125, did you replace the default moh that comes with the system? |
| 19:56.09 | jaike | run top....sometimes it eats up 99% or resources |
| 19:56.30 | jaike | asterisk 1.0.7 has problems with mp3s with id3 |
| 19:56.38 | nick125_lappy | Ariel_, yeah, i put in some different music, 64kbps, no id3, mono |
| 19:56.40 | nick125_lappy | <PROTECTED> |
| 19:56.40 | nick125_lappy | <PROTECTED> |
| 19:56.51 | jorge_bla | can i paste three lines ? |
| 19:56.57 | Nugget | yeah, the current state of MOH is pretty awful |
| 19:57.09 | Ariel_ | nick125, if you change it back to the one supplie does it still do it? |
| 19:57.15 | jorge_bla | can i paste three lines ? |
| 19:57.15 | Hmmhesays | what the hell does 'kute' mean in a text message |
| 19:57.23 | Qwell | Hmmhesays: cute? |
| 19:57.28 | jorge_bla | /home/LRP/buildroot/build_i386/staging_dir/bin/i386-linux-uclibc-gcc -o gentone gentone.c -lm |
| 19:57.28 | jorge_bla | ./gentone busy 480 620 |
| 19:57.28 | jorge_bla | make[1]: ./gentone: Command not found |
| 19:57.30 | jaike | i run a script in crond to stop mpg123 every 30 mins |
| 19:57.35 | Nugget | I'd settle for IRC that could handle unicode. |
| 19:57.37 | jaike | to make sure |
| 19:57.39 | jorge_bla | what is gentone ?? ble |
| 19:57.45 | Hmmhesays | it says 'hey kute' |
| 19:57.49 | Qwell | cutey? |
| 19:57.50 | Nugget | Cutie. |
| 19:57.51 | Hmmhesays | geebus i hate text messages |
| 19:57.55 | Qwell | however you spell it |
| 19:58.20 | pbd | Don't get me started about IRC. I was there for it's inception- a bad solution to a bad problem. |
| 19:58.22 | nick125_lappy | the highest thing in top is top |
| 19:58.27 | Hmmhesays | its like you are stuck in aol chat hell |
| 19:58.47 | Nugget | like aol chat, but more expensive. :) |
| 19:58.55 | rking | Nugget: recent xchat's (mine's 2.4.4) with properly configured locale's work great with UTF-8. ¿¡see!? |
| 19:58.57 | jaike | nick...try copying back original moh |
| 19:59.04 | Nugget | rking: that's irrelevant. |
| 19:59.13 | pbd | 15 years later, and it's gotten no better, despite thousands of programmers attempts to 'fix it'. |
| 19:59.21 | Nugget | until you convince the rest of the planet to switch to xchat, it's worthless. |
| 19:59.37 | rking | Nugget: nah, my attitude with utf-8 is "spew it and screw 'em" |
| 19:59.40 | Nugget | utf-8 is just one of a dozen "standards" for using unicode over irc. |
| 19:59.42 | jorge_bla | what is gentone? |
| 19:59.47 | Nugget | so none work, really |
| 20:00.07 | Beirdo | hey cool, there's actually an ipkg for asterisk for the wrt54g |
| 20:00.09 | Beirdo | heh |
| 20:00.11 | rking | Nugget: imho utf-8 is going to emerge at the top of the pile. what other unicode makes sense? ucs-2? |
| 20:00.17 | nick125 | i think my wireless might be going to junk |
| 20:00.18 | *** join/#asterisk FuriousGeorge (n=furious@pool-70-111-20-125.nwrk.east.verizon.net) |
| 20:00.22 | FuriousGeorge | hey all |
| 20:00.28 | nick125 | it works pretty ok on here, with a skip here and there |
| 20:00.33 | Nugget | I don't think we'll ever have a top of the pile. |
| 20:00.34 | Beirdo | this weather is lovely |
| 20:00.35 | FuriousGeorge | whats that port range im supposed to foreward on rtp to get sip to traverse nat? |
| 20:00.44 | Nugget | I think irc is and forever will be a 7-bit clean medium |
| 20:00.46 | FuriousGeorge | 3000-4000? 4000-5000? |
| 20:00.57 | jaike | 10000 - 20000? |
| 20:00.59 | pbd | George- check your sip.conf.. the range should be specified. |
| 20:01.04 | nick125 | so now i dont think its asterisk |
| 20:01.10 | FuriousGeorge | jaike: that sounds right |
| 20:01.21 | FuriousGeorge | thanks pbd |
| 20:01.59 | pbd | You can also set the range there.. but be careful- first timers typically set it too small, and not all endpoints can deal with 'nonstandard' ranges. |
| 20:02.04 | Hmmhesays | if you start ser in debug mode are all the modules supposed to echo 'initializing'? |
| 20:02.41 | Beirdo | I gave up on SER so quick :) |
| 20:02.47 | Hmmhesays | cause nathelper, tm, sl, usrloc and registrar don't. where textops stateless and maxfwd do |
| 20:02.56 | rking | Nugget: why 7-bits? i've been outputting utf-8 for a while now and it definitely seems like the servers do fine with it. word is next version of windows is going to have much better utf-8 support (and Microsoft has never broken a promise) |
| 20:03.07 | Nugget | rking: the servers don't care. |
| 20:03.14 | Nugget | but most people aren't seeing what you're typing. |
| 20:03.21 | Nugget | which is the crux of the problem |
| 20:03.25 | jaike | i think 10000-20000 is the usual range...ive set iptables to allow all udp on those ports...is there a security risk? |
| 20:03.25 | rking | Nugget: right - so it's just the clients. exactly how many clients matter? 3? |
| 20:03.28 | FuriousGeorge | definately not in sip.conf but i believe 10k-20k is correct |
| 20:03.41 | Nugget | rking: maybe 30 I'd guess. |
| 20:03.48 | pbd | Careful, now- don't mix client and server issues- that's one of the 'broken' bits of IRC- the protocol is fuzzy. |
| 20:03.52 | Beirdo | rtp.conf, I think |
| 20:03.58 | Nugget | including quite a few which are console based where any sort of unicode is a tremendous hemmorhoid. |
| 20:04.25 | jaike | beirdo: yup .. thats it |
| 20:04.31 | Beirdo | IRC and UTF-8 are not compatible. not until the servers start supporting languages properly |
| 20:04.31 | *** join/#asterisk ChibaPet (n=mason@acheron.hsd1.ma.comcast.net) |
| 20:04.50 | Beirdo | so the clients can actually negotiate support |
| 20:04.54 | ChibaPet | Hey, all. Anyone familiar with SPA-3000 want to render some insight for me? |
| 20:05.01 | pbd | George- sorry.. I should have said RTP.conf. |
| 20:05.01 | rking | Beirdo: what sort of data chokes any IRC server? (sorry, this is off-topic) |
| 20:05.01 | Beirdo | and that's not likely to ever happen |
| 20:05.04 | queuetue | I've got a A@H server running and I just converted it to be the firewall. I can connect to the server fine from my sipura, but my bv sip connection has stopped registering properly... |
| 20:05.14 | Qwell | ChibaPet: ask away |
| 20:05.16 | Beirdo | the servers don't care |
| 20:05.19 | nick125 | i wonder if it would be better if i used wavs, but, i cant get wavs to work which sucks |
| 20:05.21 | Beirdo | the clients do |
| 20:05.28 | pbd | ChibaPet- I've got one, but I'm sort of a noob with it- I got it to work, if that helps. :) |
| 20:05.46 | darkskiez | i think its rubbish that xlite can talk to sipgate but asterisk cant thru a nat, I'm looking at ethereal dumps to see why the rtp streams get lost with *. Any ideas? |
| 20:05.46 | Beirdo | and there's no way for the clients to tell the server that they don't want UTF-8 |
| 20:06.04 | Qwell | darkskiez: firewall |
| 20:06.22 | ChibaPet | I'm new to *, and I'm trying to get to the FXO on the SPA-3000, and, indeed, I have an extension to get me to the FXO, and I hear the dial tone the FXO gives me, but each digit says: |
| 20:06.24 | ChibaPet | <PROTECTED> |
| 20:06.27 | heison | ~seen sivana |
| 20:06.29 | jbot | sivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 21h 56m 56s ago, saying: 'ya'. |
| 20:06.37 | ChibaPet | Doesn't seem to actually be passing through digits. |
| 20:07.00 | darkskiez | Qwell: its a nat'd connection. I can make and send calls, but the RTP stream doesnt make it back with asterisk. for some reason it does with xlite. |
| 20:07.14 | Qwell | darkskiez: is asterisk set to nat=yes? |
| 20:07.27 | nick125 | anyone here know why wav moh isnt working? |
| 20:07.31 | ManxPower | ~docs |
| 20:07.31 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 20:07.36 | darkskiez | Qwell: i;ve tried yes and no |
| 20:07.43 | nick125 | i put a wav in there, it doesnt show up in AMP, and it doesnt play |
| 20:07.56 | ManxPower | darkskiez: If you use asterisk's nat=yes then you must DISABLE NAT SUPPORT IN THE CLIENT |
| 20:08.05 | ChibaPet | I'm also unsure of why the FXO is not picking up on inbound calls to the PSTN, so I'm game for help with either, depending on what is more interesting for folks to help with. :) |
| 20:08.07 | rking | Beirdo: that's a simple matter of converting on the client side. Channels have encodings described in their /topic's, and clients just need enough support for encoding-per-channel, and that's all assuming that all clients shouldn't just move to UTF-8. as long as everyone on every OS can view UTF-8, the clients that don't support it should just wither away |
| 20:08.14 | Nugget | "simple"? heh |
| 20:08.18 | Beirdo | not supported |
| 20:08.30 | Beirdo | and never likely will be |
| 20:08.47 | darkskiez | Manxpower: i am using asterisk as a client to sign on to sipgate in this scenario |
| 20:08.51 | Nugget | converting irc to utf-8 is about as easy as boiling the ocean. |
| 20:08.57 | bjohnson | ChibaPet: check wiki for spa 3000 forwarding trick |
| 20:09.09 | nick125 | anyone even know if asterisk supports .wavs? |
| 20:09.13 | Beirdo | Nugget: that's a quotable one :) |
| 20:09.46 | ManxPower | darkskiez: Ah, and you did the normal localnet= externip= figured out what RTP ports your SIP gatewway is using and forwarding them? |
| 20:09.54 | ChibaPet | bjohnson - read that, and it seemed interesting, but I don't mind if the SPA-3000 picks up before passing to *. I think I have the normal case configured... But of course I don't have it configured correctly! |
| 20:09.56 | ManxPower | nick125: it does. |
| 20:10.46 | ChibaPet | I have a dial-plan it should follow that should hook me into *'s start extension. But, I don't see any message from * indicating a connection attempt. |
| 20:10.56 | *** join/#asterisk darkskie1 (n=darkskie@host86-133-151-115.range86-133.btcentralplus.com) |
| 20:10.58 | nick125 | ok, so why when i put a .wav in /var/lib/asterisk/mohmp3, it doesnt play |
| 20:11.01 | darkskie1 | ManxPower: asterisk is behind the nat. I can use xlite to talk to sipgate, both receive the calls, but no audio with asterisk |
| 20:11.13 | nick125 | is there something special i have to do to make it work with wavs? |
| 20:11.22 | ChibaPet | I am, obviously, very new to all of this, so there are any number of things I might have gotten wrong. |
| 20:11.43 | jorge_bla | jesus to crosscompile asterisk is bad dream |
| 20:11.57 | ChibaPet | I *do* have connections to the outside world working, and connections between various internal extensions. |
| 20:13.01 | ManxPower | darkskie1: you did not answer my question. |
| 20:13.12 | ManxPower | nick125: no. |
| 20:13.15 | darkskie1 | ManxPower: i got disconnectred think i missed it |
| 20:13.18 | ManxPower | nick125: with is your actual ISSUE? |
| 20:13.21 | ManxPower | darkskiez: Ah, and you did the normal localnet= externip= figured out what RTP ports your SIP gatewway is using and forwarding them? |
| 20:13.49 | darkskie1 | ManxPower: I did set them, u |
| 20:13.52 | ChibaPet | Anyway, can someone point me to information about this "Attempting native bridge" stuff, and what it might mean? Web searches haven't helped as yet, and reading the source isn't my preferred solution here, although I'll dig into it if I can't get clues elsewhere. |
| 20:13.53 | ManxPower | nick125: Mosic on Hold does not support wav, only MP3 |
| 20:13.54 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 20:13.57 | darkskie1 | i cannot forward the ports unfortunately. |
| 20:14.10 | ManxPower | darkskie1: If you cannot forward ports then you cannot use Asteirsk behind nat |
| 20:14.21 | darkskie1 | ManxPower: how does xlite do it ? |
| 20:14.35 | ChibaPet | Music On Hold can support anything that can pump out the right bits... There's a voip-info page that describes supporting different formats... |
| 20:14.38 | nick125 | ManxPower: ok then, what would be the way to best settings to make mpg123 do less (and make the moh sound less skippy and such)? |
| 20:14.39 | ManxPower | darkskie1: I'm sure they do something horrible. |
| 20:14.48 | *** join/#asterisk santiago (n=santiago@63.245.86.192) |
| 20:14.53 | ManxPower | nick125: no idea? |
| 20:15.22 | ChibaPet | nick - skippy means the machine playing your MP3s is too stressed, or your network is congested... |
| 20:15.27 | *** join/#asterisk tubecat_ (n=ksoze@tidy.obscurity.org) |
| 20:15.28 | rking | nick125: you could use a format that doesn't require decoding |
| 20:15.42 | ChibaPet | nick - drop your bitrate. The output is going over a phone, after all. |
| 20:15.59 | nick125 | ChibaPet: its already 64kbps, how low should i go? |
| 20:16.02 | nick125 | 32kbps? |
| 20:16.17 | ChibaPet | is bandwidth an issue, then? voice connections are clear, or no? |
| 20:16.20 | ManxPower | nick125: you DO realize that all codecs except for ulaw and alaw distort music, right? |
| 20:16.34 | *** join/#asterisk Koshatul (n=evangeli@CPE-138-217-190-164.qld.bigpond.net.au) |
| 20:16.43 | nick125 | ManxPower: i figured there would be some distortion |
| 20:16.46 | ManxPower | nick125: 64kbps is fine |
| 20:17.00 | ManxPower | nick125: depending on the codec, there can be a lot of distortion |
| 20:17.12 | nick125 | im trying GSM atm |
| 20:17.14 | ManxPower | things like G726 are heavily optimized for voice |
| 20:17.19 | ManxPower | GSM is pretty bad. |
| 20:17.23 | ManxPower | try ulaw if you can |
| 20:17.27 | tubecat_ | i'm having trouble setting up an iax2 channel. on my server box asterisk does not listen on any port |
| 20:17.31 | ChibaPet | nick: http://www.voip-info.org/tiki-index.php?page=Asterisk+mpg123+faking+it |
| 20:17.46 | nick125 | ManxPower: i dont see a ulaw setting on x-lite :/ |
| 20:17.56 | ManxPower | nick125: aka G711 |
| 20:18.02 | ManxPower | or g711u |
| 20:18.20 | nick125 | oh ok |
| 20:18.37 | *** join/#asterisk the_devil_dont_s (n=Adam@195.26.12.229) |
| 20:18.50 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 20:18.52 | nick125 | how much b/w does G711/G711U/ULAW use? |
| 20:18.52 | tubecat_ | so my client box can't connect. any tricky reasons why asterisk won't listen? |
| 20:19.01 | *** part/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 20:19.01 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 20:19.17 | ManxPower | nick125: 64k plus overhead, about 80k total |
| 20:19.37 | ChibaPet | So, anyway, anyone have clues about my FXO issues? I'll muddle it out eventually, but would love to save some pain and aggravation. :) |
| 20:20.04 | nick125 | ManxPower: ok, so, what codec would you suggest for over the internet (pretty much with a max 1MB upload line)? |
| 20:20.08 | jorge_bla | brrrr, why asterisk doesnt have autoconf . is anybody working on it? |
| 20:20.18 | ManxPower | nick125: make it work with g711/ulaw first. |
| 20:20.25 | ChibaPet | joege - mm, Debian binary packages |
| 20:20.37 | ManxPower | jorge_bla: because nobody will comit to maintainging it forever |
| 20:21.07 | jorge_bla | ManxPower, anyway it would be nice to have it.. |
| 20:21.08 | pbd | jorge- you're asking a religious question here. |
| 20:21.15 | jorge_bla | pbd realy? |
| 20:21.21 | jorge_bla | ha ha ha |
| 20:21.26 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 20:21.38 | pbd | Most of the developers of asterisk seem to have an allergy (for good reasons- I'm neutral on this one) to autoconf. |
| 20:21.47 | ChibaPet | No help to be had. The channel is much like the docs. :P Oh well. Maybe I'll pay someone for help at some point. Later, all! |
| 20:22.03 | ManxPower | Since Asterisk only officially supports Linux, autoconf is rather silly |
| 20:22.11 | jorge_bla | pbd, i am trying to compile asterisk agains uclibc but ... |
| 20:22.21 | pbd | Frankly, aside from a very short list of dependencies, I've never had much problem compiling it- autoconf is kind of overkill. |
| 20:22.45 | brad_mssw | what should DTMF Tx method be set for, for SIP devices? I'm having an issue where if I have a command that starts with *, it is not recognized by asterisk from a certain SIP device |
| 20:23.06 | pbd | jorge- See, now you're trying to make it work outside of it's small, well defined box- no one's warranteed * for that. |
| 20:23.20 | brad_mssw | i've got options like inband,avt,info,inband+info,avt+info, and auto ... auto doesn't seem to work properly |
| 20:23.34 | pbd | brad- Sounds like you're working with a Cisco phone. :) |
| 20:23.44 | brad_mssw | pbd: Sipura 841 :/ |
| 20:23.58 | jorge_bla | do i need sound card to use asterisk ? |
| 20:24.07 | Beirdo | no |
| 20:24.11 | pbd | Well, on the cisco, I set it for avt.. and set asterisk for rfc2833. |
| 20:24.20 | pbd | That gives me out of band dtmf. |
| 20:25.05 | *** join/#asterisk Broom (i=Broom@jescobar.ayustar.net) |
| 20:25.06 | pbd | I also specifically disable inband on the phone. |
| 20:25.07 | Broom | hey all |
| 20:25.12 | *** join/#asterisk Lordy (i=steve@d040099.adsl.hansenet.de) |
| 20:25.29 | Lordy | hi all |
| 20:25.36 | brad_mssw | pbd: thanks, let me give that a try |
| 20:25.40 | pbd | brb |
| 20:25.43 | Broom | i'm having problems with incoming calls using a TE110P card, i only get this: Starting simple switch on 'Zap/12-1' |
| 20:25.45 | Broom | then the hangup |
| 20:25.50 | Broom | any ideas? |
| 20:26.01 | Qwell | Broom: turn verbose up and debug on |
| 20:26.05 | Broom | i did |
| 20:26.06 | Lordy | i would like to know if it is possible to have a ascend max dial out on a specified line for asterisk |
| 20:26.07 | Broom | that's all i get |
| 20:26.11 | Broom | i did set verbose 10 |
| 20:26.16 | Broom | set debug 10 |
| 20:26.24 | Qwell | debug isn't an int... |
| 20:26.25 | *** join/#asterisk [1]JohnJacob (n=JohnJaco@pcp0011543387pcs.mainf01.in.comcast.net) |
| 20:26.58 | Broom | Core debug was 0 and is now 10 |
| 20:27.02 | Broom | got that with that command |
| 20:27.04 | Lordy | correction: lucent max, i wanted to say :) |
| 20:29.18 | Broom | anything else i can do to verify this? |
| 20:30.24 | nick125 | i tried that faking thing, and it didnt work :/ |
| 20:34.05 | Lordy | asterisk with lucent max ? anyone ? |
| 20:34.45 | pbd | Sorry, lordy- don't have one.. |
| 20:36.09 | heison | ~seen sivana |
| 20:36.10 | jbot | sivana <n=sivana@204.14.18.187> was last seen on IRC in channel #asterisk, 3d 22h 26m 37s ago, saying: 'ya'. |
| 20:36.46 | DaPrivateer | can someone please assist me with this... how can i make the AGI dial something on the line its currently on? IE someone calls, and when it answers the phone if they press a certain option it dials back dtmf tones on the same line. |
| 20:37.33 | DaPrivateer | (this is on a digium X100P using zaptel) |
| 20:38.45 | jsmith | DaPrivateer: SAY DIGITS 1234567890 |
| 20:38.50 | DaPrivateer | thank you |
| 20:39.21 | jsmith | Oh, wait... DTMF. |
| 20:39.29 | jsmith | That'll probably just read off the numbers.... |
| 20:39.45 | Broom | nothing on my prob? |
| 20:39.58 | dudes | SendDTMF(${EXTEN}#) |
| 20:40.22 | pbd | Broom- I don't run any Digium hardware myself.. I'd guess no one does, from the channel today. |
| 20:40.23 | jero | anyone ever integrated asterisk with a toshiba pbx using a pri ? |
| 20:40.24 | shido6 | Zzzzz |
| 20:40.25 | brad_mssw | anyone have a sipura 841 phone ? |
| 20:40.28 | shido6 | I do |
| 20:40.30 | shido6 | I run digium gear |
| 20:40.33 | shido6 | whats the problem? |
| 20:40.39 | shido6 | thats all I run is digium gear |
| 20:40.40 | jsmith | pbd: I use Digium hardware... |
| 20:41.11 | jero | pbd: its being done |
| 20:41.22 | jero | pbd: but first we have to deal with both |
| 20:41.24 | pbd | Ok, so someone talk to Broom and walk him through the zaptel config. :) |
| 20:41.27 | brad_mssw | can't dial *112 or similar because the phone tries to interpret it .... trying to figure out what setting disables that |
| 20:41.43 | Broom | jsmith: you have digium hardware for incoming calls? |
| 20:41.53 | DaPrivateer | dudes - you are my hear |
| 20:41.55 | DaPrivateer | hero* |
| 20:41.57 | DaPrivateer | thank you so much |
| 20:42.27 | Ariel_ | brad_mssw, go into the advaced setting in the phone and you will see * functions delete the one you don't want and it will pass that opiton through. |
| 20:42.27 | jsmith | Broom: Yes... |
| 20:42.54 | pbd | jero- such a conservative place. Let me guess- asterisk for voicemail first, then buy handsets and migrate over? Seems to be a common path... you might find some help on the lists, a lot of people ask about it. |
| 20:43.09 | brad_mssw | Ariel_: tried that, doesn't seem to work ... especially if the command is '**' |
| 20:43.17 | Broom | jsmith: well, my problem is that when i place a call i only get this: Starting simple switch on 'Zap/16-1' |
| 20:43.19 | Broom | then the hangup |
| 20:43.27 | Broom | i have verbose to 10 |
| 20:43.29 | Broom | and debug on |
| 20:43.33 | Broom | thats the only thing i get |
| 20:43.37 | jero | pbd: not really, all VMs on *, toshiba for analog lines and * for the PRI link. then move everything to * |
| 20:43.41 | brad_mssw | Ariel_: it pretty much immediately gives a busy signal ... works from zap phones, and a UTStarcom F1000 just fine |
| 20:43.55 | *** join/#asterisk supaigtr (n=yurplsl@152.53.16.10) |
| 20:44.07 | Ariel_ | Broom, post your setting for the zaptel and zapata.conf on pastebin.ca and your dial string so we can see what you have setup. |
| 20:44.35 | Ariel_ | brad_mssw, if you look at the dialing rules you will see it only allow one * |
| 20:45.04 | brad_mssw | configurable dialing rules ... or hard rules for the phone ? |
| 20:45.55 | nick125 | anyone here know of x-lite problems when you go to press a button (such as 1) during a call? |
| 20:46.05 | pbd | jero- Unfortunately, the last time I integrated a Toshiba, it was with the help of a toshiba vendor and using FXOs for the interconnect. |
| 20:46.16 | jero | okay |
| 20:46.18 | Broom | Ariel: dialstring? |
| 20:46.24 | jero | thanks anyway pbd |
| 20:46.25 | pbd | jero- but what's the issue you're seeing? |
| 20:46.41 | jero | no issue yet, will start in a few days |
| 20:47.05 | jero | I just fear we cant replace the analog links by a pri |
| 20:47.35 | brad_mssw | oh, there's the dialplan in advanced ... heh |
| 20:48.38 | pbd | jero- you mean, you're afraid the pri won't integrate between the toshiba and *? |
| 20:48.41 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 20:48.42 | surfdue | hi |
| 20:48.46 | jero | pbd: yes |
| 20:49.00 | surfdue | i have x-lite sip phone, i dial one in our menu and i cant get the menu to go up |
| 20:49.05 | jero | pbd: I'll probably get a toshiba vendor to |
| 20:49.06 | surfdue | the sip phone dosnt show i dialed 1 |
| 20:49.09 | surfdue | but makes a noice |
| 20:49.11 | surfdue | anyone know how to help |
| 20:49.23 | pbd | Jero- Well, the Toshiba was a little funky on FXO/FXS timing, etc- but PRI is PRI.. assuming your signalling is matched, it really should be no problem. |
| 20:49.58 | jero | pbd: how will we tell the toshiba to send the voicemails to the pri ? |
| 20:50.20 | pbd | Jero- you're planning on using a multiport digium PRI card in the * box, receive PRI from the carrier there, and pass it on to the toshiba, correct? |
| 20:50.25 | Ariel_ | jero, you have to chnage the rules on the toshiba pbx for that |
| 20:51.01 | pbd | jero- Voicemail integration has a number of pitfalls- haven't seen them personally, but the MWI stuff has had more than a few askers on asterisk-users. |
| 20:52.17 | Ariel_ | you will not have any mwi nor any way to do call back via there menu. But phones kinda work. |
| 20:52.23 | *** join/#asterisk kg (n=kg@chello062179062077.chello.pl) |
| 20:52.43 | luke-jr__ | How can I determine why a call file is being ignored? |
| 20:52.52 | luke-jr__ | there doesn't seem to be anything with -vvvvvvvvvvvv |
| 20:53.03 | jero | ariel: what happens when the toshiba phone user wants to read its mail ? |
| 20:53.24 | jsmith | luke-jr__: Does it have the correct extension. Are you writing the file somewhere else, and then moving it to the outgoing directory? |
| 20:53.27 | jero | he will dial an extension that the toshiba has to forward to * |
| 20:53.45 | *** join/#asterisk RoyK (n=roy@ti211210a080-0158.bb.online.no) |
| 20:53.50 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 20:54.16 | Ariel_ | you setup a dial rule in the toshiba to send the call via the pri then asterisk will take that and send them to voiemail. |
| 20:54.20 | RoyK | hi |
| 20:54.22 | RoyK | any idea what this is about? |
| 20:54.23 | RoyK | Aug 20 00:53:37 NOTICE[5965]: rtp.c:514 ast_rtp_read: Unknown RTP codec 105 received |
| 20:54.27 | Ariel_ | you need to be able to configure both systems |
| 20:54.31 | Broom | Ariel: posted |
| 20:54.36 | bkw_ | http://www.news24.com/News24/Backpage/Crime_Court/0,,2-1343-1345_1755475,00.html |
| 20:54.49 | pbd | Royk: The other end is sending a codec that Asterisk doesn't like? |
| 20:55.00 | luke-jr__ | jsmith: yes |
| 20:55.27 | bkw_ | RoyK, I think it might be dtmf trying to use 105 instead of 101? |
| 20:55.31 | bkw_ | whats the SDP say? |
| 20:56.04 | pbd | bkw- let's hear it for little old ladies! |
| 20:56.08 | jsmith | luke-jr__: Strange... |
| 20:56.37 | jero | thanks for the infos guys |
| 20:57.19 | luke-jr__ | jsmith: http://pastebin.ca/20462 |
| 20:58.10 | Broom | Ariel: http://pastebin.ca/20461 |
| 20:58.11 | Broom | sorry |
| 20:58.39 | nick125 | any ideas on why when someone calls 7777 and presses 1, it doesnt go the the queue? |
| 20:58.40 | RoyK | bkw_: is that related to info/rfc2833-something? |
| 20:59.01 | bkw_ | where is the sdp? |
| 20:59.18 | RoyK | er... how can i find that out? |
| 20:59.22 | bkw_ | sip debug |
| 20:59.24 | bkw_ | and post it |
| 20:59.27 | bkw_ | on pastebin |
| 20:59.37 | *** join/#asterisk Equinox (n=secret@star.l93.com) |
| 20:59.44 | RoyK | bkw_: ok |
| 20:59.47 | jsmith | luke-jr__: Did you try taking out the blank lines? |
| 20:59.56 | jsmith | luke-jr__: It's just a guess |
| 21:00.00 | luke-jr__ | jsmith: no... I've had it work w/ blank lines before, tho |
| 21:00.14 | luke-jr__ | didn't help :( |
| 21:00.41 | luke-jr__ | interestingly, an old call file that used to work is also ignored |
| 21:01.16 | jsmith | Maybe you changed your outgoing directory in asterisk.conf? |
| 21:01.35 | luke-jr__ | nope |
| 21:01.45 | luke-jr__ | the files are disappearing once moved, too |
| 21:02.23 | Equinox | The "N" in NXX is 1-9? |
| 21:02.42 | pbd | Ignored as in doesn't run, or ignored as in left there, luke? |
| 21:02.48 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 21:03.12 | zedkatuf | I'm using kphone (under linux) softphone & am trying to get incoming calls working, the CLI shows me that calls are getting as far as the asterisk box from the internet.....but I keep getting "THe number you have dialled has not been recognised" on my landline....does anyone know about kphone's settings that could perhaps be borking things...? |
| 21:03.23 | pbd | Is it possible your system is having timing issues, or permissions issues between asterisk and that outgoing directory? |
| 21:04.10 | zedkatuf | (Ican make outgoing calls with no probs) |
| 21:04.59 | pbd | zed: Sounds like a dialplan issue.. can you reach that kphone from another softclient via *? |
| 21:05.01 | luke-jr__ | aha! |
| 21:05.13 | luke-jr__ | jsmith: my asterisk doesn't run as root anymore ^^;; |
| 21:05.44 | *** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net) |
| 21:05.54 | pbd | Luke- Glad to be of help, that will be $50. ;-) |
| 21:06.16 | surfdue | does anyone use x-lite |
| 21:06.18 | surfdue | or used or know how to useit |
| 21:06.22 | surfdue | Im having a problem, when i dial an extention after im on the phone xlite dosnt seem to send it through |
| 21:06.30 | zedkatuf | pbd: good idea re another client.......I'll try that |
| 21:06.31 | surfdue | its not sending keys |
| 21:07.05 | SpaceBass | surfdue: try changing your dmtf settings |
| 21:07.28 | bkw_ | Equinox, N is 2-9 |
| 21:07.32 | surfdue | how |
| 21:07.32 | bkw_ | Z = 1-0 |
| 21:07.35 | surfdue | whats dmtf? |
| 21:07.36 | bkw_ | X = 0-9 |
| 21:07.36 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 21:07.50 | nick125 | surfdue: in the advance settings menu |
| 21:07.52 | bkw_ | er z=1-9 |
| 21:07.53 | nick125 | then dmtf |
| 21:07.57 | surfdue | where? |
| 21:07.58 | surfdue | how |
| 21:08.12 | surfdue | k |
| 21:08.17 | nick125 | surfdue: go into the settings, then go to the advanced settings menu |
| 21:08.18 | Qwell | bkw_: are there any that match ABCD? |
| 21:08.21 | nick125 | tell me what it has |
| 21:08.22 | surfdue | change 2 what? |
| 21:08.26 | surfdue | force bands is no |
| 21:08.32 | bkw_ | Qwell, A B C or D ninny :P |
| 21:08.34 | RoyK | grr |
| 21:08.34 | RoyK | Aug 20 01:08:56 WARNING[5965]: chan_zap.c:2131 pri_find_dchan: No D-channels available! Using Primary on channel anyway 202! |
| 21:08.38 | RoyK | any idea what that might be? |
| 21:08.39 | Qwell | bkw_: I mean any and all |
| 21:08.39 | surfdue | 2800 and 110 |
| 21:08.42 | surfdue | 101* |
| 21:08.46 | nick125 | hmm |
| 21:08.52 | bkw_ | RoyK, I get that message all the time |
| 21:08.53 | Qwell | (besides [abcd]) |
| 21:08.56 | bkw_ | when I first start the box |
| 21:09.04 | nick125 | thats the same as my settings |
| 21:09.12 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net) |
| 21:09.22 | SpaceBass | surfdue: you may want to check the voip-info wiki and search for dmtf to learn about the options |
| 21:09.23 | RoyK | bkw_: this is repeating. B chans 1 through, say, 12 comes up and then i get that |
| 21:09.52 | bkw_ | RoyK, its more retarted code |
| 21:10.06 | Equinox | Anyone here familiar with Polycom IP500 phones? I'm trying to set up my FTP directory and have a question: Bootrom 2.6.2 has a bootrom.ld file.. But no bootrom.ver .. The 2.6.1 file has both.. Do I not need the .ver? |
| 21:10.12 | bkw_ | :P |
| 21:10.16 | brad_mssw | ok, fixed the dialplan on my sipura 841 phone. One issue though, whenever Authenticate() is called, it does not appear to work from this phone ... the # sign does nothing ... is this a dtmf code thing I need to trial and error ? extension numbers work fine |
| 21:10.31 | RoyK | bkw_: but then it doesn't mean much? |
| 21:10.49 | surfdue | omg |
| 21:10.55 | surfdue | can someone just tell me what to do. |
| 21:10.57 | surfdue | lol |
| 21:10.58 | RoyK | <PROTECTED> |
| 21:10.58 | RoyK | Aug 20 01:11:17 WARNING[5965]: chan_zap.c:2131 pri_find_dchan: No D-channels available! Using Primary on channel anyway 202! |
| 21:10.58 | RoyK | <PROTECTED> |
| 21:11.01 | RoyK | that sort of thing |
| 21:11.08 | SpaceBass | surfdue: read |
| 21:11.11 | surfdue | it works on other clients |
| 21:11.16 | surfdue | just not mine |
| 21:11.19 | surfdue | on nicks it works |
| 21:11.21 | surfdue | not mine? |
| 21:11.29 | bkw_ | RoyK, I see that all the time.. the span goes down and back up |
| 21:11.30 | surfdue | nick what does ure dfm say |
| 21:11.37 | RoyK | span == dchan? |
| 21:11.41 | bkw_ | but only on start/restart |
| 21:11.45 | pbd | brad: sounds definitely like a dtmf issue- I've been in that particular ring of hell lately. |
| 21:11.53 | bkw_ | what is channel 202? |
| 21:11.58 | bkw_ | does it line up with a dchan? |
| 21:11.58 | RoyK | bkw_: this happens all the time, not while restarting |
| 21:12.03 | pbd | brad: make sure that your dtmf settings are matched all around. |
| 21:12.13 | nick125 | surfdue: same |
| 21:12.20 | RoyK | bkw_: dhcan on span 7 |
| 21:12.47 | *** part/#asterisk darkskie1 (n=darkskie@host86-133-151-115.range86-133.btcentralplus.com) |
| 21:12.54 | PupenoL | can't asterisk be 'configured' te use a prefix (like /usr/local) when building it ? |
| 21:12.54 | surfdue | whats magic numbers |
| 21:12.59 | brad_mssw | pbd: k, thanks. |
| 21:12.59 | surfdue | im in gsm config |
| 21:13.08 | surfdue | nick get on skype |
| 21:13.12 | SpaceBass | no magic, just needs to match |
| 21:13.33 | surfdue | so the gsm number needs to match the dtfm? |
| 21:13.51 | SpaceBass | http://www.google.com/search?client=safari&rls=en&q=xlite+dtmf+asterisk&ie=UTF-8&oe=UTF-8 |
| 21:14.32 | surfdue | what do i do |
| 21:14.51 | SpaceBass | anyone see that slashdot article today about GSM phone integration? |
| 21:14.56 | gambolputty | ywa |
| 21:15.02 | gambolputty | article didnt mention asterisk |
| 21:15.06 | gambolputty | just the slashdot thing |
| 21:15.26 | *** join/#asterisk nick125 (n=nick125@unaffiliated/nick125) |
| 21:15.26 | jaike | finally found out how to make polycom volume permanent! yes! dont need to buy new headsets |
| 21:15.53 | SpaceBass | there is specifically an article on asterisk and gsm integration |
| 21:16.03 | surfdue | http://www.google.com/search?client=safari&rls=en&q=xlite+dtmf+asterisk&ie=UTF-8&oe=UTF-8 |
| 21:16.05 | Qwell | not much "article" to id |
| 21:16.07 | Qwell | it* |
| 21:16.27 | RoyK | bkw_: how should i debug this? will a pri debug give much hunch? |
| 21:16.28 | surfdue | anyone? |
| 21:16.29 | surfdue | please |
| 21:16.37 | surfdue | how do i fix this sip x-lite problem |
| 21:16.52 | surfdue | not sending the keys i click one calling someone |
| 21:17.14 | SpaceBass | dude, seriously... you need to research a little... there is no simple answer, it depends on how your extension is set up and what the settings in xlite are, etc |
| 21:18.04 | Nivex | jaike: How? My friend just got some Polycoms and is having that problem. |
| 21:18.31 | opus_ | yeah has ANYBODY built a gsm cellphone network to asterisk intergration |
| 21:18.37 | opus_ | what a load of fucking crap |
| 21:18.41 | SpaceBass | lol |
| 21:18.59 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 21:19.01 | zoa | probably someone did |
| 21:19.09 | zoa | but it would have been a cellphone carrier |
| 21:19.18 | DrmCtchr | where is the simple switch context defined at ? |
| 21:19.23 | surfdue | SpaceBass, maybe u can go over it with me |
| 21:19.25 | SpaceBass | there are mini-cells out there.... |
| 21:19.29 | surfdue | i relly just want to get xlite wokring |
| 21:19.32 | opus_ | i think that the transporter from star trek is a great idea to |
| 21:19.40 | opus_ | wheres my flying fucking car |
| 21:19.42 | surfdue | in amp does this matter dtmfmode: |
| 21:19.50 | DrmCtchr | i need to have dialtone played there and preferably wait a bitlonger for digits from the t1 E&M channels |
| 21:19.54 | Qwell | surfdue: very much so, yes |
| 21:20.03 | SpaceBass | surfdue: absloutly... thats the point :) |
| 21:20.11 | surfdue | ok well 2 extentions |
| 21:20.18 | surfdue | have the same crap in it dtmfmode: rfc2833 |
| 21:20.19 | SpaceBass | surfdue: what ever it says in AMP for your extension, is what xlite should be set to |
| 21:20.25 | nick125 | surfdue: the extensions are the same other then the password and username |
| 21:20.27 | SpaceBass | try changing it to inband |
| 21:20.41 | surfdue | how |
| 21:20.49 | surfdue | nvm |
| 21:20.57 | *** part/#asterisk jsmith (n=jsmith@64.50.35.114.2O7.net) |
| 21:21.12 | brad_mssw | pbd: yep, setting the dtmf equal on both sides fixed it ... also set it to use ulaw instead of gsm ... thanks |
| 21:21.38 | DrmCtchr | anyone know where is the simple switch context defined at ? |
| 21:21.41 | zoa | out of band should be better if you ask me |
| 21:21.48 | Broom | hey, now i'm getting this error: Spawn extension (from-trunk, s, 2) exited non-zero on 'Zap/7-1' |
| 21:21.54 | *** join/#asterisk Gronker__ (n=Gronker2@70.152.167.143) |
| 21:22.25 | opus_ | zoe unless you type in your bank account number |
| 21:22.33 | surfdue | nope |
| 21:22.34 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 21:22.37 | surfdue | dosnt work |
| 21:22.56 | surfdue | dammiot |
| 21:23.24 | SpaceBass | try another client |
| 21:23.33 | SpaceBass | firefly can use IAX2 and works well |
| 21:23.40 | surfdue | link me please |
| 21:23.46 | SpaceBass | iaxcomm on linux or os x... |
| 21:23.48 | Qwell | ~google firefly iax2 |
| 21:23.49 | surfdue | anyclient that works, works for me |
| 21:25.45 | Equinox | Do the polycom config files contain the sip user & password? |
| 21:26.23 | pbd | brad- glad to be of assistance. |
| 21:27.23 | pbd | zoa- If you can't use Ulaw, out of band is the *only* way to go. |
| 21:28.28 | *** join/#asterisk yellowsnow (n=yellowsn@dsl-213-134-245-123.solcon.nl) |
| 21:30.29 | *** join/#asterisk ryansc (n=ryansc@c-67-164-188-180.hsd1.co.comcast.net) |
| 21:30.30 | adelas | hey, i have my asterisk@home setup, and cisco phones setup already, what kind of info would i need to add a trunk (i have a mediatrix 1204 setup) |
| 21:31.18 | Broom | anyone can help me with this error: Spawn extension (from-pstn-reghours, s, 3) exited non-zero on 'Zap/12-1' |
| 21:31.20 | adelas | well, i'm kinda loat in amp |
| 21:31.24 | adelas | lost* |
| 21:32.05 | adelas | i have the mediatrix ip and so on, just don't know how to set it up |
| 21:32.08 | pbd | Err.. check priority 3 in your s extension in the fron-pstn-reghours context? ;-) |
| 21:32.31 | opus_ | does anyone have the problem wioth the IP 501 when a user dials *98 it actually modifies it and makes it 9*8? |
| 21:32.51 | opus_ | whats there to change? I forget what the correct dial pattern modification is.. |
| 21:32.58 | opus_ | if somebody could cut and paste it, would be cool |
| 21:34.36 | surfdue | x-lite not sending numbers i dial |
| 21:34.39 | surfdue | can anyone help |
| 21:36.48 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 21:37.02 | bkw_ | WHO IS THAT AT MARKS DESK |
| 21:37.16 | file | bkw_: guess who just called sales |
| 21:37.26 | DrmCtchr | please, anyone know where is the simple switch context defined at ? |
| 21:37.29 | MikeJ[Laptop] | bill gates? |
| 21:37.44 | surfdue | i cant find a sollution |
| 21:37.47 | surfdue | ive searched every |
| 21:37.48 | surfdue | wehre |
| 21:37.58 | surfdue | i cant find a reason why it wont send numbers once im on the phone |
| 21:38.12 | surfdue | if im in voicemail it sayspush one to say voicemail, i push one nothing happens |
| 21:38.18 | surfdue | it acts as if it was never pushed |
| 21:38.20 | surfdue | can anyone help |
| 21:38.31 | MikeJ[Laptop] | file, well who already? |
| 21:38.41 | file | MikeJ[Laptop]: it was some woman who was answering an ad for a job that pays $25/hour |
| 21:38.44 | *** join/#asterisk DaPrivateer (n=matt7229@gateway.teamfloco.com) |
| 21:39.00 | file | how the hell she got me I have no clue |
| 21:39.30 | DaPrivateer | Ok, general question directed towards anyone in this channel running asterisk in a production environment. I am trying to convince my boss to let me transfer our phone system over to VoIP. What can you tell me about your uptime percentage? |
| 21:39.34 | MikeJ[Laptop] | heh! |
| 21:39.51 | MikeJ[Laptop] | DaPrivateer, depends. |
| 21:40.04 | puowvip | 100%, but I only have 7 extensions. |
| 21:40.13 | puowvip | since April 2005. |
| 21:40.20 | DaPrivateer | ya, we have only 4 extensions (all off-site) and 3 PSTN lines |
| 21:40.25 | opus_ | daprivateer if you loose registration sometimes it doesn't connect back. |
| 21:40.56 | DaPrivateer | MikeJ[Laptop] what does it depend on? |
| 21:40.59 | opus_ | daprivateer its about the same quaility of a cell phone. sometimes doesn't work. i say 1/20 calls won't go through just because its so complicated. |
| 21:41.06 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 21:41.08 | MikeJ[Laptop] | depends on what you are doing... |
| 21:41.28 | surfdue | someone? |
| 21:41.28 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 21:41.30 | surfdue | PLEASE! |
| 21:41.34 | surfdue | if anyone can help |
| 21:41.44 | Qwell | if anybody knew, they'd help |
| 21:41.49 | MikeJ[Laptop] | asterisk in loads like that, doing only tdm to voip and nothing else should be pretty solid |
| 21:41.49 | DaPrivateer | MikeJ[Laptop] - just giving offsite personnel access to PSTN lines in the office |
| 21:41.52 | Qwell | OR, you could pay for help |
| 21:42.02 | *** join/#asterisk blitzrage (n=leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:42.03 | DaPrivateer | ok; when does it cause problems |
| 21:42.06 | Qwell | don't, however, beg. it makes you look stupid |
| 21:42.08 | MikeJ[Laptop] | but your uptime will be very dependant on your lines. |
| 21:42.36 | surfdue | well i am stupid |
| 21:42.40 | surfdue | else i would have fixedit. |
| 21:42.42 | DaPrivateer | ya, clearly. i meant the server itself though. clearly if the internet goes down, thats not asterisk's fault |
| 21:42.43 | Qwell | then RTFM |
| 21:42.47 | Qwell | ~docs |
| 21:42.47 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 21:43.01 | opus_ | get a brain moran |
| 21:43.10 | Qwell | a brain, or some cash |
| 21:43.12 | MikeJ[Laptop] | DaPrivateer, like any software, the more complex the functions you use, the more problems you have.. it depends... people using complex proxy solutions, or using rarely used functions have the best luck finding bugs |
| 21:43.23 | MikeJ[Laptop] | but for what you are talking about, asterisk should be very solid |
| 21:43.24 | DaPrivateer | hehehehe |
| 21:43.53 | DaPrivateer | the only other thing in it will be IAX to FWD for toll free calls |
| 21:44.13 | MikeJ[Laptop] | yeah, you are talking about the easiest of functionality... |
| 21:44.27 | blitzrage | Qwell: do you know if Asterisk accepts registrations from multiple interfaces on the same box? (I think it should... but apparently not on one of my boxes - running, 1.0.9) |
| 21:44.35 | MikeJ[Laptop] | with no transcoding, you could run that on a wrt |
| 21:45.06 | MikeJ[Laptop] | blitzrage, iax? sip? |
| 21:45.27 | blitzrage | MikeJ[Laptop]: IAX2 |
| 21:45.29 | MikeJ[Laptop] | the issue to my understanding is on virtual interfaces |
| 21:45.39 | MikeJ[Laptop] | but seperate physical is supposed to be ok |
| 21:45.41 | blitzrage | MikeJ[Laptop]: yah, these aren't virtual - physical |
| 21:45.44 | Qwell | virtual interfaces are pointless anyhow |
| 21:45.52 | blitzrage | MikeJ[Laptop]: hrmmm... I have a device that won't register for some reason |
| 21:45.53 | Qwell | just give your interface two IPs, heh |
| 21:46.00 | MikeJ[Laptop] | what's it do? |
| 21:46.09 | MikeJ[Laptop] | bindaddr set? |
| 21:46.16 | blitzrage | MikeJ[Laptop]: not entirely sure... bindaddr not set |
| 21:46.28 | Qwell | blitzrage: check netstat, make sure its listening |
| 21:46.30 | blitzrage | MikeJ[Laptop]: by default does it just bind to the first address if 0.0.0.0 is not set? |
| 21:46.46 | MikeJ[Laptop] | I don't recall.. I don't multihome. |
| 21:46.49 | *** join/#asterisk durak (n=durak196@85.98.97.122) |
| 21:46.54 | DaPrivateer | lol.. anyone else? my boss wants more people saying it works well lol |
| 21:47.06 | blitzrage | DaPrivateer: it works well |
| 21:47.10 | Qwell | asterisk? |
| 21:47.11 | Qwell | no, it sucks |
| 21:47.13 | Qwell | :P |
| 21:47.15 | DaPrivateer | hehe |
| 21:47.16 | blitzrage | yah, its shit |
| 21:47.23 | DaPrivateer | uptime percentage? |
| 21:47.29 | blitzrage | 2% |
| 21:47.29 | Qwell | like 3%? |
| 21:47.37 | Qwell | yeah, its about 2.5 |
| 21:47.42 | blitzrage | Qwell: wow! thats awesome |
| 21:47.45 | DaPrivateer | heh, seriously plz |
| 21:47.45 | Qwell | (blitzrage: nice one, btw) |
| 21:48.06 | surfdue | We are interconnected with the PSTN with a CISCO and we use RFC 2833 for DTMF because most part of the GWs in the network work much better than In Band. |
| 21:48.07 | surfdue | Therefore customers that use XPRO have to configure DTMF force send in Band as No. |
| 21:48.07 | surfdue | CISCO recognizes XPRO is not sending DTMFs in band but for some reason ignores all DTMFs sent via XPRO. |
| 21:48.07 | surfdue | All the rest, (PAP, Micronet, ZOOM, etc) gateways work fine on RFC 2833, but XPRO. |
| 21:48.07 | surfdue | Is this a bug? |
| 21:48.14 | surfdue | THis guy has the same problem as me? |
| 21:48.58 | surfdue | i tryed dtmf to yes and no on force still dosnt seem to work |
| 21:49.12 | DaPrivateer | surfdue - i would say contact xpro |
| 21:49.19 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 21:49.26 | blitzrage | DaPrivateer: with Asterisk, anyone can tell you all sorts of things, but you REALLY have to go and do it yourself and see if it'll work for you - it requires testing, it requires knowledge. Asterisk is for your TELEPHONES - you only get as much out of them as you put into them. |
| 21:49.44 | Qwell | eww |
| 21:49.45 | surfdue | i have x-lite |
| 21:49.48 | Qwell | perv |
| 21:50.02 | Qwell | better |
| 21:50.06 | blitzrage | lol |
| 21:50.09 | DaPrivateer | blitzrage - ya i have a PBX set up right now and i love it. thanks to help from some people here i even got around the problems i was having |
| 21:50.21 | DaPrivateer | he just wants to see other people's success stories basically |
| 21:50.21 | blitzrage | back latah |
| 21:50.44 | MikeJ[Laptop] | DaPrivateer, the amount of people in here should say somthing |
| 21:50.50 | MikeJ[Laptop] | there are lots of people using it |
| 21:50.52 | DaPrivateer | ya, thats what i said |
| 21:50.54 | Qwell | MikeJ[Laptop]: heh |
| 21:50.58 | Qwell | should say something, which way? ;] |
| 21:51.08 | Qwell | alot of people == alot of people that need help |
| 21:51.15 | Qwell | the question is, whats the ratio? |
| 21:51.20 | Hmmhesays | only luv you gets from a tv set and it makes you crazay |
| 21:52.05 | *** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) |
| 21:52.25 | DrmCtchr | where is the simple switch configuration defined at ? |
| 21:53.14 | DaPrivateer | thanks guys |
| 21:53.18 | file | eh? |
| 21:54.51 | *** join/#asterisk queuetue (n=queuetue@h69-21-252-54.69-21.unk.tds.net) |
| 21:55.09 | puowvip | ok |
| 21:55.16 | Kraven | hrm it seems that noone is looking on their bugtracker @ inaccess |
| 21:55.26 | Kraven | :( |
| 21:55.40 | Druken | file: could have been worse, you could be encrypted too :) |
| 21:56.14 | *** part/#asterisk secure75 (n=mic@gfwlan.cablesurf.de) |
| 21:57.02 | vonage | ow |
| 21:57.05 | vonage | so i have to ask |
| 21:57.11 | vonage | why does everyone want to be me |
| 21:57.12 | vonage | im me |
| 21:57.14 | vonage | knock it off |
| 21:57.20 | Qwell | nobody wants to be vonage |
| 21:57.26 | Druken | hell no |
| 21:57.39 | vonage | n |
| 21:57.51 | Equinox | Anyone here offer some advise on polycom config files? |
| 21:57.55 | Hmmhesays | vonage does have a decent network put together |
| 21:57.59 | Druken | i think i'd agree with Qwell |
| 21:58.09 | vonage | that i do |
| 21:58.10 | brad_mssw | nope, we get all circuits busy messages on vonage all the time |
| 21:58.11 | Hmmhesays | anyone who says vO naaaaajjjjj should be shot |
| 21:58.18 | vonage | yeah im a busy guy |
| 21:58.23 | brad_mssw | and we can't pull the lines into asterisk without going through these damn linksys boxes first |
| 21:58.27 | Hmmhesays | i never get all circuits busy on the business plan |
| 21:58.34 | Hmmhesays | brad_mssw: you are mistaken |
| 21:58.37 | brad_mssw | this is a business plan |
| 21:58.38 | vonage | yeah pay more call more i like to say |
| 21:58.44 | *** join/#asterisk wpbrown (i=wpbrown@66.0.163.142) |
| 21:58.46 | Hmmhesays | because I have all my lines going through asterisk |
| 21:58.47 | brad_mssw | we've got 4 business lines |
| 21:58.48 | surfdue | hmm |
| 21:58.54 | brad_mssw | Hmmhesays: all of them? how? |
| 21:58.59 | surfdue | i wish tehre was a quik fix to this DTMF settings error thing |
| 21:58.59 | brad_mssw | Hmmhesays: via softphone stuff? |
| 21:59.09 | Hmmhesays | hell no, i have my asterisk registered directly with vonage |
| 21:59.09 | *** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net) |
| 21:59.13 | brad_mssw | Hmmhesays: vonage says the softphone accounts can't do call hunt which we need |
| 21:59.25 | vonage | hey give me time give me time |
| 21:59.27 | brad_mssw | Hmmhesays: ok, i'm confused, how? what login info do you have to use? |
| 21:59.34 | Hmmhesays | the login info they provided me |
| 21:59.42 | brad_mssw | Hmmhesays: are you in the US? |
| 21:59.44 | Hmmhesays | indeed |
| 21:59.52 | file | hot'n'sexy Hmmhesays! |
| 21:59.54 | brad_mssw | Hmmhesays: hmm, and you're on the standard business phones? |
| 21:59.55 | Hmmhesays | yo |
| 22:00.00 | Hmmhesays | business plus |
| 22:00.05 | puowvip | nice nick, Hmmhesays |
| 22:00.08 | rabelais | has anyone sucessfully connected to earthlink's new "free calling online" sip service? despite all my attempts, the server still rejects my login...I can't register with the sip server and I have no idea why not |
| 22:00.12 | Hmmhesays | yo file wassup |
| 22:00.16 | brad_mssw | Hmmhesays: yeah, ok ... hmm ... do you have a contact with vonage to get that info? |
| 22:00.26 | file | oh nothing, getting zaptel setup on a box |
| 22:00.27 | brad_mssw | Hmmhesays: or did you just call their standard support lines ? |
| 22:00.33 | Hmmhesays | if you order the business plus plan they will give it to you |
| 22:00.37 | vonage | you could ask me ^_^ |
| 22:00.40 | vonage | except not |
| 22:00.58 | Hmmhesays | because if you order it in the midwest you will probably be talking to me |
| 22:01.20 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 22:01.22 | Hmmhesays | i dunno bout their other business plans yet |
| 22:01.34 | Hmmhesays | residential definately not |
| 22:01.38 | brad_mssw | Hmmhesays: we're on the Small Business Unlimited |
| 22:01.44 | brad_mssw | Hmmhesays: is that the same |
| 22:01.48 | Hmmhesays | nope |
| 22:02.13 | Hmmhesays | how much you paying a month for that? |
| 22:02.24 | brad_mssw | Hmmhesays: $50/mo each line |
| 22:02.26 | Hmmhesays | ouch |
| 22:02.33 | brad_mssw | Hmmhesays: so like $200 total + our 800 number |
| 22:02.48 | Qwell | just get a per minute account with a real provider, heh |
| 22:02.49 | brad_mssw | Hmmhesays: so they have different business plans then ?? |
| 22:02.56 | Hmmhesays | the smallest business plus plan is 4 simultaneous calls with 5000k minutes us/canada for $150 a month |
| 22:03.02 | Hmmhesays | and you can use asterisk |
| 22:03.19 | brad_mssw | Hmmhesays: that's not bad |
| 22:03.25 | Hmmhesays | nope, works great too |
| 22:03.26 | brad_mssw | Hmmhesays: is this something you have to call about ? |
| 22:03.32 | brad_mssw | Hmmhesays: or do they have a business website ? |
| 22:03.34 | Hmmhesays | official launch is in 2 weeks |
| 22:03.41 | brad_mssw | ah hah |
| 22:03.48 | Hmmhesays | soft launch was 2-3 months ago |
| 22:03.51 | brad_mssw | ok, we'll probably be switching to that |
| 22:03.54 | Equinox | Hmmhesays- 5 million minutes? |
| 22:03.55 | Hmmhesays | you can get it right now |
| 22:03.58 | Hmmhesays | oops |
| 22:03.59 | Hmmhesays | 5k |
| 22:04.05 | Hmmhesays | 5000 minutes is what I meant |
| 22:04.09 | Qwell | 5k minutes maximum? |
| 22:04.10 | Equinox | Ahh :) |
| 22:04.14 | Equinox | That's 3c/minute |
| 22:04.23 | brad_mssw | yeah, converting from digital -> analog -> digital probably isn't the best thing here ;) |
| 22:04.46 | Hmmhesays | just call them and ask for the business plus info |
| 22:04.53 | brad_mssw | cool, thanks |
| 22:04.56 | Hmmhesays | if you're in the midwest you'll get sent here |
| 22:05.00 | Qwell | damn thats expensive, heh |
| 22:05.24 | Hmmhesays | Qwell that would be expensive if you didn't get 4 did's with it |
| 22:05.24 | brad_mssw | now, when you say 4 simultaneous calls, how does that get configured in asterisk ... as 4 individual register lines ? |
| 22:05.31 | Hmmhesays | I use one |
| 22:05.33 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-45.west.biz.rr.com) |
| 22:05.37 | DrmCtchr | anyone have a sounds file of dialtone? |
| 22:05.40 | Hmmhesays | and have the phones in a ring group |
| 22:06.02 | mover | when i compile head from cvs i got a depend error in order i copied it from one server to another and try to do a make |
| 22:06.06 | mover | what happen here? |
| 22:06.16 | mover | need i do a make xxxx ? |
| 22:06.17 | brad_mssw | Hmmhesays: are they fax capable by any chance? |
| 22:06.22 | DrmCtchr | or know of anyway to have asterisk play dialtone ? |
| 22:06.27 | *** join/#asterisk santiago (n=santiago@63.245.87.180) |
| 22:06.36 | Qwell | DrmCtchr: playtones(dialtone) |
| 22:06.38 | Hmmhesays | you can try to fax on it, but like any voip provider chances are you'll get fax failure |
| 22:06.38 | Qwell | erm, dial |
| 22:06.42 | Qwell | and, playtone |
| 22:06.54 | DrmCtchr | sweet thanks |
| 22:07.03 | Qwell | no, it is playtones |
| 22:07.06 | brad_mssw | Hmmhesays: well, they advertise the fax stuff on their website (and we do use it via the linksys device) ... seems to work fine |
| 22:07.15 | brad_mssw | Hmmhesays: what codec do they use, btw ? |
| 22:07.17 | Hmmhesays | nod, if you got the bandwidth |
| 22:07.20 | brad_mssw | Hmmhesays: ulaw ? |
| 22:07.29 | Hmmhesays | they'll use whatever you want, but preferred is g711 on their network |
| 22:07.44 | brad_mssw | ok, cool |
| 22:07.59 | Hmmhesays | let me rephrase, they'll use g711 alaw/ulaw and g729 |
| 22:08.11 | Hmmhesays | basically whatever cisco does they'll use |
| 22:08.12 | Equinox | hat is g729? |
| 22:08.14 | brad_mssw | is g729 lossy ? |
| 22:08.16 | Hmmhesays | yes |
| 22:08.22 | Qwell | aren't all codecs? |
| 22:08.27 | Hmmhesays | yes |
| 22:08.32 | Hmmhesays | g711 is uncompressed though |
| 22:08.32 | Equinox | flac isn't lossy ;) |
| 22:08.34 | mover | can anyone gimme e hint please? |
| 22:08.38 | Equinox | NOt that it's used in voip |
| 22:08.40 | Qwell | Equinox: nor is flac a codec ;p |
| 22:08.43 | brad_mssw | Hmmhesays: g711 is uncompressed ? |
| 22:08.44 | adelas | hey, why cant i make a phone call on my cisco phone, its connected to the aah box i know |
| 22:08.48 | Hmmhesays | brad_mssw: yes |
| 22:08.48 | Qwell | a telephony codec anyhow |
| 22:08.49 | Equinox | Qwell- Well it could be ;) |
| 22:08.51 | adelas | i have the trunk setup, and so is outbond setup |
| 22:08.54 | brad_mssw | didn't realize that ... hmm |
| 22:09.10 | Hmmhesays | although the sampling rate on it is shit |
| 22:09.13 | adelas | am i missing something here? |
| 22:09.53 | adelas | its like my asteirks box is not communicating with my phone/internet line |
| 22:10.03 | adelas | i did a fdw setup to test it out first |
| 22:10.16 | Hmmhesays | drop me a line if you want any more info brad_mssw i'm out of here for now |
| 22:10.20 | mover | i got all times a make: *** [.depend] Error 2 |
| 22:10.38 | brad_mssw | Hmmhesays: thanks ... I may take you up on that when we switch over ;) |
| 22:10.42 | *** join/#asterisk jkitchen (i=kitchen@tarnishing.microsofts.name) |
| 22:10.46 | Hmmhesays | later |
| 22:10.56 | jkitchen | anyone having issues with broadvoice today? |
| 22:11.12 | jkitchen | my dialplan was working fine with them yesterday.. and today my asterisk server keeps sending 404 |
| 22:12.39 | rabelais | jkitchen: today and almost everyday...just doesn't work |
| 22:12.51 | jkitchen | it's like they changed something though |
| 22:13.06 | jkitchen | because if I create an extension in my default context with my phone number.. it works |
| 22:13.21 | rabelais | agreed...calls used to actually terminate to my box, today I'm just getting a busy signal if I dial in |
| 22:13.24 | jkitchen | but i have been using this thing for the past several days without such a thing |
| 22:13.49 | jkitchen | so your machine worked yesterday, but not today? |
| 22:14.00 | jkitchen | you should sip debug and see if you're getting a 404 as well |
| 22:14.03 | rabelais | well, I don't know...I don't think I got any calls in yesterday from that number |
| 22:14.06 | jkitchen | i keep getting sent to voicemail |
| 22:14.14 | jkitchen | 'busy' voicemail |
| 22:14.19 | jkitchen | maybe they did screw something up |
| 22:14.24 | rabelais | I have voicemail turned off...so I don't know |
| 22:14.26 | jkitchen | yea |
| 22:14.59 | jkitchen | their website is really slow today as well |
| 22:15.14 | SkramX | hello all. |
| 22:15.37 | jkitchen | rabelais: can you do me a favor? |
| 22:15.40 | rabelais | I take it back...that call actually does get to my * box... |
| 22:15.46 | jkitchen | hrm |
| 22:15.49 | jkitchen | dammit |
| 22:15.53 | rabelais | but after I send the ack, it dies |
| 22:15.58 | rabelais | no response from them |
| 22:17.23 | rabelais | jkitchen: do yourself a favor and look elsewhere for a provider...I struggled with them for over 8 months before I finally said screw it |
| 22:17.49 | rabelais | the only reason I still have my account is to hang onto my number...I'm in the last stages of a number transfer and I'll finally be free of them |
| 22:17.58 | jkitchen | Looking for 7144632069 in from-bv |
| 22:17.58 | jkitchen | Reliably Transmitting (no NAT): |
| 22:17.58 | jkitchen | SIP/2.0 404 Not Found |
| 22:18.06 | jkitchen | heh |
| 22:18.21 | jkitchen | know of any asterisk-friendly providers? |
| 22:18.35 | rabelais | as in iax? |
| 22:18.42 | rabelais | or just asterisk friendly? |
| 22:18.45 | jkitchen | iax would be nice, but not necessary |
| 22:18.51 | jkitchen | just... more asterisk friendly than say vonage |
| 22:18.51 | rabelais | I went to telasip |
| 22:18.52 | jkitchen | heh |
| 22:19.25 | rabelais | seems to be only one guy running the show, name's gene...he talks alot, but he's nice...and the service is stable |
| 22:19.47 | mishehu | bah. |
| 22:19.56 | rabelais | the only problem I have is that I'm on the west coast, and a quirk about media path redirection hasn't been worked out yet... |
| 22:20.34 | jkitchen | i'm on the west coast as well |
| 22:20.35 | *** join/#asterisk SpaceBass (n=SpaceBas@c-24-125-184-203.hsd1.va.comcast.net) |
| 22:20.36 | rabelais | so sometimes call quality gets weird during heavy traffic...we've been working on figuring it out, I'll be really happy when that issue is solved |
| 22:20.59 | jkitchen | grrr |
| 22:21.25 | jkitchen | this thing is pissing me off... i'm new to asterisk.. here I think I have it almost working.. and now I can't get it working at all |
| 22:21.33 | zedkatuf | jkitchen |
| 22:21.35 | zedkatuf | Hi |
| 22:21.41 | jkitchen | hi. |
| 22:21.45 | zedkatuf | I found myself typing grrr on another channel at same time |
| 22:21.50 | jkitchen | heh |
| 22:21.51 | zedkatuf | I'm new to * as well |
| 22:21.53 | zedkatuf | :) |
| 22:22.06 | zedkatuf | I'm just an ebd user who wants to ditch his landline |
| 22:22.23 | zedkatuf | so my issues are somewhat simpler than a lot of folks |
| 22:22.28 | jkitchen | my boss wants automated call handling |
| 22:22.33 | jkitchen | so i'm setting up an asterisk box |
| 22:22.38 | rabelais | jkitchen: if you want to test things out, setup a fwd account for yourself as well, that'll help you determine where things are funky with your configs versus broadvoice's general flakyness |
| 22:22.44 | zedkatuf | ..yeah, that's more tricky... |
| 22:22.55 | jkitchen | zedkatuf: i had the dialplan working |
| 22:22.59 | zedkatuf | re testing: I've got outbound calling working |
| 22:23.01 | jkitchen | but today.. I called .... and got nothing |
| 22:23.05 | zedkatuf | wierd |
| 22:23.16 | *** join/#asterisk trash_ (i=trash@databerlin.org) |
| 22:23.18 | zedkatuf | try another ITSP mebbe |
| 22:23.20 | jkitchen | yea outbound calling still works for me |
| 22:23.30 | pc4 | How do you fix the good old they can hear me but I can't hear them syndrome? |
| 22:23.32 | pc4 | =) |
| 22:23.47 | zedkatuf | possibly oopen up more ports on ur firewall |
| 22:23.49 | jkitchen | outbound calling works perfectly, in fact |
| 22:23.59 | jkitchen | inbound calling was working, perfectly, yesterday |
| 22:24.00 | pc4 | zedkatuf - What am I missing? I opened 5060 incoming. |
| 22:24.08 | jkitchen | today my * keeps giving a 404 |
| 22:24.17 | zedkatuf | 5060 & 5061 UDP |
| 22:24.25 | *** join/#asterisk bkw__ (n=brian@adsl-69-155-21-122.dsl.tulsok.swbell.net) |
| 22:24.25 | *** mode/#asterisk [+o bkw__] by ChanServ |
| 22:24.28 | zedkatuf | 10000 - 20000 UDP |
| 22:24.34 | zedkatuf | possibly also |
| 22:24.42 | nick125_lappy | anyone here know of a good sip client for linux other then x-lite that works with asterisk? |
| 22:24.43 | zedkatuf | 3478 - 3479 UDP & TCP |
| 22:24.51 | zedkatuf | kphone works gr8 |
| 22:24.55 | pc4 | zedkatuf - Incoming or only outgoing? |
| 22:24.57 | DrmCtchr | there is no tones called dialtone |
| 22:24.58 | jkitchen | nick125_lappy: kphone seems to work.. but x-lite works awesome |
| 22:25.25 | SpaceBass | nick125: iaxcomm |
| 22:25.28 | rabelais | nick125: I've heard good things about sflphone... |
| 22:25.29 | nick125_lappy | well, my friend is having a problem with x-lite on his computer, and we cant seem to get it working right :/ |
| 22:25.36 | rabelais | I just can't get it to compile ;P |
| 22:25.38 | SpaceBass | gnomconf or what ever works too |
| 22:25.45 | jkitchen | gnome-meeting ?> |
| 22:25.48 | SpaceBass | thats the onew |
| 22:25.51 | jkitchen | :) |
| 22:25.54 | *** join/#asterisk tmx (n=tomx@dot.snat.baz.cz) |
| 22:25.56 | jkitchen | yea, I don't run gnome though |
| 22:26.05 | jkitchen | kde 4tw |
| 22:26.14 | SpaceBass | iaxcomm |
| 22:26.20 | nick125_lappy | i think its dtmf problems, but, his settings are exactly the same as mine for dtmf |
| 22:26.38 | zedkatuf | nick125_lappy: |
| 22:26.45 | zedkatuf | xlite probs under linux or winblows? |
| 22:26.50 | nick125_lappy | hes in linux |
| 22:27.55 | nick125_lappy | so am me |
| 22:28.00 | nick125_lappy | so am i |
| 22:28.01 | nick125_lappy | bleh |
| 22:28.03 | SpaceBass | is he on the same lan as the * box? |
| 22:28.09 | nick125_lappy | no, its over the internet |
| 22:28.22 | SpaceBass | switch to iaxcomm use iax2... make your lives easier :) |
| 22:28.30 | nick125_lappy | it connects right tothe asterisk, voice works, it just wont send when he presses a button |
| 22:28.33 | nick125_lappy | is that a gui app? |
| 22:28.43 | SpaceBass | yeah |
| 22:29.14 | nick125_lappy | looks good |
| 22:29.38 | rabelais | does gnomemeeting support sip? |
| 22:29.44 | SpaceBass | not sure why my IRC client decided to make that an action.... |
| 22:29.52 | SpaceBass | rabelais: i believe so |
| 22:29.55 | *** join/#asterisk jsaunders (i=js@70.70.74.152) |
| 22:30.05 | zedkatuf | nick125_lappy: I tried xlite under linux....the sound is not quite working properly atm....using kphone works gr8 however |
| 22:30.07 | SpaceBass | rabelais: i see it supported a lot |
| 22:30.18 | jsaunders | What are they main ways of passing dtmf... inband, rfc2833, any others? |
| 22:30.19 | SpaceBass | kphone... forgot about that one |
| 22:30.22 | nick125_lappy | zedkatuf, i cant find out how to put the password in there |
| 22:30.31 | zedkatuf | in kphone? |
| 22:30.38 | nick125_lappy | yeah |
| 22:30.53 | zedkatuf | are u trying to register through asterisk? |
| 22:31.08 | nick125_lappy | im trying to register kphone on my asterisk box, yeah |
| 22:31.29 | SpaceBass | outband |
| 22:31.50 | zedkatuf | nick125_lappy: ok, username: is extension number (eg 500) |
| 22:31.59 | zedkatuf | when u click on Register button |
| 22:32.10 | zedkatuf | it'll pop up with a box for u 2 type pword |
| 22:32.14 | nick125_lappy | it will ask for the pass then? |
| 22:32.16 | nick125_lappy | ok |
| 22:32.46 | nick125_lappy | what about gnome meeting, anyone use that? |
| 22:32.47 | ard | qwlekjqwe |
| 22:32.51 | ard | sorry |
| 22:34.04 | SpaceBass | nick125_lappy i have |
| 22:34.07 | SpaceBass | been a while |
| 22:34.18 | SpaceBass | is your name nick? |
| 22:34.28 | nick125_lappy | yes |
| 22:34.34 | jkitchen | my name is kitchen ;( |
| 22:34.38 | jkitchen | some foo has my nick tho |
| 22:34.50 | pc4 | sip:1234@asterisk -- which one is the username/pass? sip/1234? |
| 22:36.15 | SpaceBass | actually the computer is osx2.nsnet.com but the user that auto loggs in is kitchen |
| 22:37.38 | Equinox | Anyone using polycom sip 1.5.x? |
| 22:38.30 | nick125 | there we go |
| 22:38.34 | pc4 | Has anyone hooked a cell phone to their * box for free calls with mobile to mobile minutes? |
| 22:39.01 | *** join/#asterisk Himeko (n=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 22:39.07 | *** join/#asterisk nicox (n=nicox@h082218027030.host.wavenet.at) |
| 22:39.32 | nick125 | pc4: that would be interesting to see |
| 22:39.38 | nicox | hello guys, is there anybody in the channel who compiled asterisk-cvs-head in the last week? |
| 22:39.50 | SpaceBass | pc4: www.slashdot.org |
| 22:39.54 | SpaceBass | article on that right now |
| 22:40.23 | pbd | No, I think he's talking something slightly different. |
| 22:40.31 | pc4 | SpaceBass - similar article, but he was talking about setting up his own cell tower =) |
| 22:40.35 | pbd | You want to use your cell phone as an outgoing trunk from asterisk.. |
| 22:40.39 | jkitchen | hrm. |
| 22:40.48 | pc4 | pbd - Incoming trunk, out through cheap-o voip |
| 22:40.52 | SpaceBass | pc4: yeah, but read the first coment... there is a thing called voiceblue or something that does what you are talking about |
| 22:41.06 | pc4 | Well, it is a device that interfaces any cell phone into a pots, yes |
| 22:41.08 | pbd | in the slashdot article, they're setting up a gsm cell local to your house, and gatewaying the cell station out through VoIP so you don't use the minutes. |
| 22:41.12 | pc4 | so I suppose you could make your own dialplan and do that. |
| 22:41.19 | SpaceBass | i've got a sprint pcs connection card... thought about trying to integrate it once... but im not ready to write linux voice drivers for it :) |
| 22:41.20 | pc4 | But I was wondering if anyone has ever done it... and how bad is it. |
| 22:41.30 | pc4 | SpaceBass - Are they free voice minutes? |
| 22:41.31 | SwK[Work] | hah |
| 22:41.45 | SpaceBass | pc4: mine is unlimited voice and data |
| 22:41.46 | pbd | I've not looked at it much, but chan_bluetooth may do what you need. |
| 22:41.58 | SpaceBass | its not a bluetooth device :( |
| 22:42.01 | SwK[Work] | "we're not getting your dtmf" which "c=IN IP4 255.255.255.255" in the sdp for the rtp host |
| 22:42.33 | nicox | has anyone a compiling problem with cvs-head also? |
| 22:42.36 | pbd | Well then, there are some makers that allow you to plug your cell phone in and use it as an outbound device for a regular cordless phone.. something could probably be built from that with an FXS card. |
| 22:42.45 | SpaceBass | yeah |
| 22:42.53 | pbd | nico- I did last night, until I realized I had a munged Makefile. |
| 22:43.35 | nicox | how can i solve the problem? |
| 22:44.01 | nicox | everytime i try to compile it hangs |
| 22:44.04 | pbd | Depends on your problem, exactly. You could try re-checking out head to a new directory, and copying over the Makefile- but my problem isn't necessarily yours. |
| 22:44.26 | wunderkin | nicox: check ps |
| 22:44.31 | pbd | I blew up my compile because the makefile I had somehow didn't have a reference to one of the more critical and new source files. |
| 22:45.08 | nicox | hm, i heared the fourth time that cvs-head makes troubles |
| 22:46.04 | *** join/#asterisk schwank (n=schwank@tidy.obscurity.org) |
| 22:46.14 | nicox | i tried to download it i think 20 time ... |
| 22:46.21 | schwank | how do I make a caller enter into voicemail after a timeout on a queue? |
| 22:47.14 | wunderkin | nicox: oh i just saw your post on the ml, maybe see if it gave you any warnings about your version of bison? |
| 22:47.32 | file | ZX81 is a crazy nut |
| 22:47.33 | schwank | any takers? |
| 22:47.35 | shido6 | dialplan logic |
| 22:47.37 | shido6 | t,1 |
| 22:48.06 | shido6 | u have a s,1 in the context there? |
| 22:48.12 | schwank | shido6, then how to I have it go to different voicemails? |
| 22:48.14 | schwank | yes |
| 22:48.20 | schwank | I have 5 different queues |
| 22:48.21 | schwank | though |
| 22:48.29 | schwank | in the same context |
| 22:48.30 | nick125_lappy | anyone know of a voip place that offers something maybe like 10 minutes free pstn or something so i can test my asterisk? |
| 22:48.38 | schwank | and I'd like them to go to different voicemails on fail |
| 22:48.47 | schwank | (I require internationalization) |
| 22:48.53 | wunderkin | nick125, voipjet gives you 0.25 |
| 22:49.29 | *** join/#asterisk ZX81 (n=ZX81@222-153-118-124.jetstream.xtra.co.nz) |
| 22:49.44 | ZX81 | ok it's running again now |
| 22:49.47 | ZX81 | call me back |
| 22:49.48 | ZX81 | :) |
| 22:50.01 | file | ZX81: :P |
| 22:50.10 | ZX81 | very strange - that registration was supposed to be to fwd |
| 22:51.38 | schwank | hmm? |
| 22:52.02 | nick125 | wunderkin: thanks |
| 22:52.37 | Druken | fwd... i have one of those... i think.... |
| 22:52.59 | *** join/#asterisk Blissex (n=Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 22:53.36 | hardwire | I should maybe donate b/w-cpu to voip-info.org |
| 22:53.39 | *** part/#asterisk ZX81 (n=ZX81@222-153-118-124.jetstream.xtra.co.nz) |
| 22:54.10 | nick125 | ok i got a voipjet account, now how do i set it up with amp? |
| 22:54.19 | Druken | hardwire: you use the wiki that much ? |
| 22:54.24 | hardwire | yes |
| 22:54.26 | hardwire | its like.. on me |
| 22:54.33 | nick125 | is it a IAX trunk? |
| 22:54.33 | nick125 | lol |
| 22:54.41 | nicox | does anybody know about a problem to compile cvs-head? |
| 22:55.05 | Druken | i must admit... i use the wiki often myself... not so much as before |
| 22:56.06 | hardwire | well |
| 22:56.08 | hardwire | its my reference |
| 22:56.23 | Druken | and what a reference it is.... |
| 22:57.30 | Druken | hardwire: you have any experince with the international calls? |
| 22:59.18 | pc4 | Can anyone test a sip server for me? |
| 22:59.29 | *** join/#asterisk buddho (n=buddho@host86-133-210-171.range86-133.btcentralplus.com) |
| 23:01.38 | *** join/#asterisk surfdue_ (n=surfdue@user-0c6t1g9.cable.mindspring.com) |
| 23:02.48 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 23:04.26 | pc4 | Can anyone test a sip server for me? please? =) |
| 23:08.10 | Druken | pc4: for? |
| 23:08.21 | hardwire | I should learn to just offline it |
| 23:08.21 | hardwire | maybe thats what I will do |
| 23:08.21 | hardwire | wget ahoy |
| 23:08.21 | hardwire | mut: mutilator: hows the life? |
| 23:09.13 | hardwire | Druken: newp |
| 23:09.43 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:09.43 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 23:10.01 | zoa | hey drunken sailer |
| 23:11.28 | pc4 | Druken - Just make a phone call on it. |
| 23:11.36 | *** join/#asterisk alexhopper (i=Alex@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 23:11.46 | pc4 | Druken - I'm not receiving audio from it. |
| 23:11.57 | hardwire | I had sushi |
| 23:11.59 | hardwire | my mind is racing |
| 23:12.04 | hardwire | I think its the coffee. |
| 23:12.26 | Druken | pc4 check your rtp |
| 23:12.49 | schwank | oh. |
| 23:13.00 | schwank | it wasn't working because modules.conf had noload app_voicemail... |
| 23:13.00 | schwank | :\ |
| 23:13.30 | pc4 | Druken - I don't know where to start :( -- I think it might be my firewall at work -- if I pm you some credentials, can you test it? |
| 23:14.08 | hardwire | whats the sip fromuser? |
| 23:14.10 | hardwire | vs CID? |
| 23:14.45 | Druken | pc4: i'll guarentee it's your firewall at work :) |
| 23:14.54 | pc4 | Druken - IT could be the firewall on the server too =) |
| 23:14.57 | pc4 | Druken - CAn you test it? :) |
| 23:14.58 | zedkatuf | re modules.conf, should Icomemtn one of these out.....: |
| 23:14.58 | zedkatuf | noload => chan_alsa.so |
| 23:14.59 | zedkatuf | noload => chan_oss.s |
| 23:15.15 | Druken | pc4: well... it could also be BOTH :) |
| 23:15.15 | zedkatuf | Icomomtn = I comment |
| 23:15.35 | zedkatuf | pc4: has Druken managed to test? |
| 23:16.11 | surfdue_ | does anyone here know alot of HTMF, using the X-Lite client? |
| 23:16.32 | zedkatuf | soz no idea :( |
| 23:16.34 | pc4 | zedkatuf - Nope :( |
| 23:16.38 | nick125 | grr |
| 23:16.43 | pc4 | Can anyone test a sip server for me? please? =) |
| 23:16.48 | zedkatuf | psate me the settings again pc4 |
| 23:17.02 | nick125 | every time i try to call out, i get 'all circuits busy, try again later' |
| 23:17.27 | zedkatuf | poss firewall prob nick125 (?) |
| 23:18.20 | nick125 | how would i check if the firewall is blocking it? |
| 23:18.24 | *** part/#asterisk santiago (n=santiago@63.245.87.180) |
| 23:18.38 | nick125 | ok |
| 23:18.46 | zedkatuf | nick125: does ur firewall have logs? |
| 23:18.48 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 23:18.49 | surfdue | sorry |
| 23:18.57 | nick125 | i need someone to help me set up the truck and outbound in AMp for voipjet |
| 23:18.58 | surfdue | If anyone replied please repeat |
| 23:19.02 | nick125 | zedkatuf: hold on |
| 23:20.30 | nick125 | zedkatuf: ok |
| 23:20.33 | nick125 | what should i look for? |
| 23:20.58 | DrmCtchr | how come there is such a big delay between: -- Starting simple switch on 'Zap/48-1' |
| 23:21.01 | DrmCtchr | and : -- Executing Answer("Zap/48-1", "") in new stack |
| 23:21.10 | surfdue | Anyone, if they know dtmf or X-LIte please tell me i really need help getting dtmf working |
| 23:22.24 | DrmCtchr | is there a configuration for simple swithc im missing? |
| 23:24.24 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 23:25.43 | zedkatuf | nick125: um, not sure really..depends on urfwall I guess |
| 23:27.13 | *** join/#asterisk netsurfer (n=bbjunkie@i-83-67-48-18.freedom2surf.net) |
| 23:27.18 | *** join/#asterisk Moc (n=mochouin@modemcable203.101-70-69.mc.videotron.ca) |
| 23:27.20 | Druken | if you bring a call in on a E1, do you have to do anything special to toss it back out the E1 ? |
| 23:27.26 | Moc | vacation are over ... |
| 23:27.45 | MikeJ[Laptop] | Druken, dial? |
| 23:28.03 | Druken | well, aside from that... |
| 23:28.18 | Druken | does it carry the original DID inside the stream somewhere? |
| 23:28.29 | MikeJ[Laptop] | not sure what your asking |
| 23:28.36 | nick125 | will this work as the dial plan: 9|1. |
| 23:28.50 | *** join/#asterisk Jzalae (n=sk@bb-205-209-93-139.gwi.net) |
| 23:28.52 | nick125 | i want to dial out like this: 9<area><num> |
| 23:29.17 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:29.23 | nick125 | without dialing the 1 |
| 23:31.51 | *** join/#asterisk SexyKen (n=ksandell@c-67-161-5-149.hsd1.ca.comcast.net) |
| 23:31.56 | SexyKen | Anyone know what this error means: 0819171852|sip |4|00|Registration failed User: 1207, Error Code:480 Temporarily not available |
| 23:32.12 | MikeJ[Laptop] | nick125, _9XXXXXXXXXX,n,Dial(1${EXTEN:1}) |
| 23:34.45 | nick125 | anyone have voipjet setup with AMP? |
| 23:36.10 | Ariel_ | nick125, yes |
| 23:36.24 | Ariel_ | works |
| 23:36.47 | nick125 | like to lend a litle assistance? :) |
| 23:37.00 | nick125 | i setup the trunk like this pretty much: |
| 23:37.43 | *** part/#asterisk tubecat_ (n=ksoze@tidy.obscurity.org) |
| 23:37.48 | nick125 | IAX2 trunk |
| 23:38.15 | nick125 | outbound caller id: "sometexthere" <1234>, Maximum Channels: Blank |
| 23:38.44 | nick125 | oh i think i might have found the issue |
| 23:38.58 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 23:39.16 | nick125 | nope |
| 23:39.17 | nick125 | ok |
| 23:39.37 | nick125 | Dialing rules: 1XXXXXXXXXX |
| 23:39.49 | nick125 | Outbound Dialing Prefix: blank |
| 23:39.58 | nick125 | trunk name: voipjet |
| 23:40.12 | nick125 | peer details: (the long thing voipjet gives out) |
| 23:40.20 | nick125 | incoming is blank |
| 23:40.29 | nick125 | register string: blank |
| 23:40.36 | nick125 | whats wrong with that? |
| 23:40.49 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-6.cybersurf.com) |
| 23:41.36 | Druken | nick125: prefix would be 9 |
| 23:41.44 | Druken | dialing rules would be XXXXXXXXXX |
| 23:42.10 | Druken | would allow 97055551212 |
| 23:42.28 | jarrod | anyone gotten reinvite working with SER? |
| 23:42.39 | DrmCtchr | how come there is such a big delay between: -- Starting simple switch on 'Zap/48-1' |
| 23:42.40 | pc4 | How can I see what password an incoming sip user is sending? |
| 23:42.54 | DrmCtchr | and the first thing in the icoming context |
| 23:43.11 | nick125 | Druken: but, in the outbound rules, it takes the 9 as the outgoing prefix |
| 23:43.37 | Druken | nick125: oh |
| 23:43.55 | Druken | then yeah, blank, and 9NXXXXXXXXX |
| 23:44.31 | Ariel_ | nick125, don't put nothing but the phone number with voipjet they don't deal well with names |
| 23:45.17 | jarrod | bah.. ser is not allowing reinvite to occur between this sipura and my cisco gateway |
| 23:45.53 | MikeJ[Laptop] | pc4, sip debug |
| 23:46.11 | file[laptop] | oh say can you "Segmentation fault." |
| 23:46.28 | MikeJ[Laptop] | yes |
| 23:46.30 | MikeJ[Laptop] | I can |
| 23:46.52 | MikeJ[Laptop] | I just did... |
| 23:47.00 | file[laptop] | I seg faulted a few times today |
| 23:47.01 | file[laptop] | silly code |
| 23:47.17 | MikeJ[Laptop] | your code, or other stuff? |
| 23:47.21 | *** join/#asterisk roche (n=roche@obiwan.inalambrica.net) |
| 23:47.23 | file[laptop] | other stuff |
| 23:47.29 | MikeJ[Laptop] | what'd you find? |
| 23:47.36 | file[laptop] | ICD |
| 23:47.39 | MikeJ[Laptop] | hehe |
| 23:48.33 | nick125 | Ariel_: ok |
| 23:49.27 | nick125 | it still says all circuits are busy |
| 23:49.50 | roche | Hi People, sometimes my asterisk show this error "chan_zap.c:5830 ss_thread: CallerID returned with error on channel " what could be ? |
| 23:49.53 | *** join/#asterisk da_monumental_1 (n=da_monum@rrcs-24-172-102-198.midsouth.biz.rr.com) |
| 23:51.28 | nick125 | Ariel_: what did you put for register string? |
| 23:51.39 | pc4 | MikeJ[Laptop] - Where does it show the password though? |
| 23:51.40 | adelas | anyone here use a mediatrix? |
| 23:52.41 | *** join/#asterisk apardo (n=w0w0@125.Red-83-46-188.pooles.rima-tde.net) |
| 23:52.54 | *** join/#asterisk SwK (n=SwK@12-219-144-126.client.mchsi.com) |
| 23:59.09 | surfdue | does anyone know dtmf |
| 23:59.14 | surfdue | or x-lite sip |
| 23:59.57 | MikeJ[Laptop] | dual tone multi frequency? |
