| 00:00.05 | theblue | For example, "Happy Birthday" is illegal if you perform it in public without paying ASCAP. |
| 00:00.06 | justinu | RIAA is the real evil |
| 00:00.06 | pbd | Hmm. Sticky on that one. If it's off-air, it's free to all who have an antenna in the US. But if you rebroadcast it, it's a problem. |
| 00:00.11 | Hymie | what the heck is ASCAP anyhow.. there's an organization in Canada that handles these things for artists, but it's not called ASCAP |
| 00:00.29 | kippi1 | is there away to test your music on hold? |
| 00:00.39 | Hymie | theblue: thaqt's performing though, not rebroadcasting |
| 00:00.51 | theblue | Hymie: Either way, its still copyright infringement./ |
| 00:00.57 | Hymie | no, not either way |
| 00:01.00 | pbd | Actually, the rules in the US say that if the copyright has expired (17 years or so), you can rebroadcast all you want. So most MOH sources use either specifically licensed stuff, or dead composers (classical recordings)- ASCAP doesn't care about those. |
| 00:01.02 | theblue | Hymie: Oh yes it is. |
| 00:01.07 | Hymie | again, it's legal to rebroadcast here in Canada |
| 00:01.21 | Hymie | you can't modify the signal |
| 00:01.31 | justinu | are you required to add "eh" at the end of every sentence? |
| 00:01.40 | oliverqg | pbd: are u from guatemala? |
| 00:01.46 | pbd | hymie- hate to see a US lawyer take THAT one to court. You've modified the signal- you're compressing it with g.729. |
| 00:02.09 | justinu | extra life? |
| 00:02.10 | Hymie | the cable companies can not. but they do not need to pay a single penny.. as long as it is an off of air, rebroadcast over cable.. and the law is loosely enough written, so apartment buildings can do the same same.. and on hold music as well |
| 00:02.11 | oliverqg | anyone a clue for my GET VARIABLE issue??? |
| 00:02.17 | pbd | oliver: No, just using them (possibly unfairly) as a state in which US copyright law (or any law) is tough to enforce. |
| 00:02.29 | oliverqg | true |
| 00:02.37 | Hymie | pbd: who cares if its a US lawyer, they have to play by Canadian legislation once on Canadian soil |
| 00:02.59 | justinu | i thought canada had adopted international copyright law treaties |
| 00:02.59 | FuriousGeorge | Hymie: you canadians are way ahead with the laws up there |
| 00:03.15 | theblue | Hymie: I love the way Canadian law works. |
| 00:03.27 | pbd | YES! Disconnect tone detection works for Brazil now! I oWn Telephonica. ;-) |
| 00:03.28 | FuriousGeorge | second only to dutch law |
| 00:03.31 | generalhan | does anyone know if its possible to extend the VM Box limit higher than 100 ? |
| 00:03.40 | Hymie | justinu: yes.. we did.. about what.. 50 years ago.. even so, adopting to a treaty does not mean all laws are identical, just that specific points are |
| 00:03.48 | justinu | ok |
| 00:03.53 | theblue | Anything's possible in Asterisk. That's how it got its name. |
| 00:03.58 | generalhan | lol |
| 00:04.04 | justinu | i dunno, i'm having a few issues with dial macros |
| 00:04.14 | pbd | Hymie- All lawyers do is enforce what's written in the law- but no law is clear, it's all open to human interpretation. |
| 00:04.17 | generalhan | ok let me rephrase, can some one explain to me how to extend the VM Boxes to hold more than 100 messages ? |
| 00:04.28 | FuriousGeorge | theblue: i never thought about it that way. i always thought of a telephone's button |
| 00:04.45 | pbd | generalhan: Shoot the users who want to store more than 100 messages? |
| 00:04.50 | generalhan | lol |
| 00:04.54 | theblue | FuriousGeorge: Right. |
| 00:04.57 | justinu | maybe it's just a #define |
| 00:04.58 | Hymie | pbd: you bet it is.. howver, this law has a long history of precidence.. including cable companies taking off air signals and converting them to digital and transmitting them to homes |
| 00:05.11 | theblue | FuriousGeorge: * is the UNIX wildcard, meaning "anything". That's the official way Asterisk got its name. |
| 00:05.21 | generalhan | no its a line set aside for people we dont want to talk to, so i push them into a VM box that i have someone check each day, but im pushing so many people in there now-a-days that it fills up too fast |
| 00:05.33 | pbd | Who can keep track of more than 100 messages stored in VM? Is it an answering machine, or a toll quality (ick) digital jukebox? |
| 00:05.34 | FuriousGeorge | theblue: it was also the msdos wildcard ;) |
| 00:05.37 | FuriousGeorge | *is |
| 00:05.37 | justinu | i need to figure out how to interrupt a dial macro when the dialed party disconnects |
| 00:05.39 | theblue | FuriousGeorge: True. |
| 00:05.46 | FuriousGeorge | dir /s *.wav |
| 00:05.47 | justinu | it seems like all my playback commands and stuff just keep running |
| 00:06.06 | theblue | FuriousGeorge: Indeed, it is. Most OSes and programming languages use * as a wildcard. |
| 00:06.07 | pbd | Hymie: Translation- no one has fought them hard enough yet. ;-) |
| 00:06.14 | Hymie | pbd: keep in mind that the way TV stations up here can "force" a cable company to pay for their feed, is by not doing air broadcasts.. which our superstations do, and so on |
| 00:06.16 | queuetue | How do I find the digium service where they will record messages for you? |
| 00:06.27 | justinu | * is the pointer deref operator in C/C++ |
| 00:06.36 | pbd | http://thevoice.digium.com |
| 00:06.38 | xheliox | http://thevoice.digium.com |
| 00:06.43 | pbd | beat yah! |
| 00:06.49 | xheliox | :( |
| 00:07.04 | *** join/#asterisk ^X-works (n=r0x0r@host111-109.pool8257.interbusiness.it) |
| 00:07.09 | generalhan | there isnt a default recording for "200" lol so if i could make it store 200 messages i wonder what would happen when the voice is suposed to say you have 200 messages ? |
| 00:07.13 | queuetue | Is it off the air, or is safari acting up? |
| 00:07.24 | pbd | Hymie- the US has it's quirks too, btw. According to the providers, no household in the US has more than one cable box. |
| 00:07.44 | Hymie | heh |
| 00:07.56 | pbd | Because they charge extra per outlet in your house. So do the cable companies. But there is no way for the providers to audit the delivery side- so... |
| 00:08.00 | Hymie | yet mysteriously peple have two or three tvs ;) |
| 00:08.28 | pbd | Once I worked for MTV (seriously), and learned that, I had no issues with splitting my cable myself to multiple TV's. |
| 00:08.35 | Hymie | heh, for sure |
| 00:08.49 | theblue | pbd: You worked for MTV? |
| 00:08.53 | Hymie | speaking of MTV, they're trying to make another attempt at coming up here |
| 00:08.55 | pbd | If Comcast wants to take issue with it, I'll ask them to show me where they're paying more for it. :) |
| 00:08.55 | queuetue | Hrm... does digium.com block canadian visitors? |
| 00:09.02 | Hymie | their first attempt failed pitifully |
| 00:09.11 | Hymie | queuetue: not that I've seen |
| 00:09.12 | pbd | I did. I worked for MTV Networks many moons ago, in their IT department. |
| 00:09.19 | theblue | pbd: MUST....KILL! |
| 00:09.28 | queuetue | Hymie: Is digium.com off the air for you? |
| 00:09.41 | justinu | works for me |
| 00:09.50 | pbd | It was a short consulting gig- they canned the entire department about 3 months after I got there. |
| 00:10.00 | Hymie | queuetue: nope |
| 00:10.11 | queuetue | Hrm... Then why can't I get there? |
| 00:10.12 | Hymie | pbd: heh, what did you do ;) |
| 00:10.23 | Hymie | queuetue: temporary routing issues between your ISP and them |
| 00:10.28 | Hymie | likely ;) |
| 00:10.30 | theblue | pbd: Oh, nevermind then. |
| 00:10.34 | theblue | Is IAXTel at all reliable? |
| 00:10.44 | pbd | Hymie: Sad truth? Three months of reading the BA's documentation, meeting with users, and playing solitaire. The app never got off the ground. |
| 00:10.57 | justinu | lol |
| 00:10.58 | Hymie | "Hi, my name is pbd, and I singlehandedly managed to get 23 people fired!" |
| 00:11.03 | Hymie | pbd: doh |
| 00:11.05 | justinu | lol again |
| 00:11.08 | pbd | But it was cool to see MTV from the inside. |
| 00:11.17 | theblue | pbd: MTV is eeeeevil. |
| 00:11.28 | justinu | agreed |
| 00:11.40 | pbd | I was actually *TOLD* To play solitaire, by my boss. We were told- don't do anything, we're not sure if we're going to continue the project.. for 2 months. |
| 00:11.57 | pbd | Nice way to make a living. |
| 00:12.02 | pbd | But.. BORING. |
| 00:12.03 | file | oh god, that made me laugh Hymie |
| 00:12.16 | theblue | pbd: Oh god. |
| 00:12.17 | queuetue | I'd find a job and quit, I think... |
| 00:12.35 | justinu | i'd play a better game than solitaire |
| 00:12.43 | pbd | That, and walking past the militant muslims on times square, shouting out 'Death to all white people', while standing next to marine recruiters.. there were times I wondered if I would make it to the subway. :) |
| 00:13.37 | pbd | What a country! |
| 00:13.55 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 00:14.28 | theblue | Ok, then. |
| 00:15.16 | theblue | For killing chat, you shall be punished. |
| 00:15.31 | theblue | I sentence you to helping me get my sip.conf, iax.conf, and extensions.conf going. |
| 00:15.40 | justinu | easy |
| 00:15.58 | Hymie | whomever said "USE THGE H EXTENSION STUPID!" ... thanks ;) it's working |
| 00:16.08 | pbd | Simple.. 'cd /etc/asterisk ; rm *.conf ;'. On a clear disk, you can seek forever. |
| 00:16.14 | theblue | ... |
| 00:16.44 | pbd | Too late. |
| 00:16.55 | netsurfer | someone already whacked him I think :P |
| 00:17.05 | pbd | Anyway, guys- you can now continue to harrass me behind my back. I'm off to home and bed. |
| 00:17.18 | justinu | later |
| 00:17.24 | generalhan | have fun pbd |
| 00:17.59 | pbd | later, guys. |
| 00:17.59 | theblue | Night, pbd. |
| 00:18.10 | generalhan | can anyone help me figure out how to make my VM Box hold 150 messages instead of 100 ? |
| 00:18.33 | *** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 00:18.40 | cio | Hi all. How's CVS feeling today? |
| 00:18.43 | justinu | just ask it |
| 00:19.08 | xheliox | working ok for me |
| 00:19.41 | cio | Cool. |
| 00:20.15 | Hymie | hmm |
| 00:20.18 | Ariel_ | did not know CVS had feelings |
| 00:20.28 | Hymie | is there an extension that gets called, when a call is answered and a bridge is formed? |
| 00:20.41 | Hymie | like the 'h' for hangup? |
| 00:20.47 | justinu | you're working on the same kind of stuff I am |
| 00:20.54 | justinu | sounds like it, at least |
| 00:21.16 | Hymie | I'm doing dual dial.. one to my cell and another to my internal extension |
| 00:21.38 | file | Hymie: no, no there isn't |
| 00:21.44 | Hymie | but, when calling my cell, I don't want to answer the external number.. as it could be the cell doing a "no answer" transfer to my asterisk box.. as it does so for voicemail |
| 00:21.50 | Hymie | hmm |
| 00:22.01 | Hymie | so, when a dial is successful, you just have to wait until it times out :( |
| 00:22.07 | Hymie | er, is completed |
| 00:22.12 | file | yes, they're bridged together |
| 00:22.17 | Hymie | is there a "time out" extension |
| 00:22.20 | file | two things can't happen at once like that |
| 00:22.35 | Hymie | like "run this in 60 seconds, now go do other stuff"? |
| 00:22.47 | Hymie | timeout command |
| 00:22.48 | Hymie | that is |
| 00:22.50 | Hymie | timed command |
| 00:22.52 | justinu | i'd like to be able to dial out to multiple devices, then based on which one answers, bridge the call |
| 00:22.52 | Hymie | whatever it may be ;) |
| 00:22.56 | file | not really, no |
| 00:23.00 | Hymie | doh |
| 00:23.07 | file | you can't do two things at once on the same channel... |
| 00:23.15 | Hymie | but I want to ;) |
| 00:23.16 | *** join/#asterisk konfuzed (n=konfuzed@H129.C72.B0.tor.eicat.ca) |
| 00:23.19 | file | too bad |
| 00:23.22 | Ariel_ | justinu, dial(sip/123&sip/234&Sip/234) |
| 00:23.29 | Hymie | yes, that works |
| 00:23.30 | *** join/#asterisk Micc (n=dotirc@c-67-171-6-249.hsd1.wa.comcast.net) |
| 00:23.31 | justinu | yeah, i got that |
| 00:23.40 | justinu | but see, I want to do different stuff based on which one answers |
| 00:23.47 | Hymie | ah |
| 00:23.49 | justinu | like if sip/123 answers, bridge the call |
| 00:23.50 | Ariel_ | justinu, then use a macro |
| 00:24.04 | justinu | but if sip/234 answers, ask it a bunch of questions, and only if it responds right, bridge the call |
| 00:24.10 | justinu | i don't think it's possible even with macros |
| 00:24.13 | marc324 | i have x100p and asterisk, whats basic test to check everything works? |
| 00:24.27 | marc324 | simple test |
| 00:24.32 | Ariel_ | plug a phone line to it and dial into it or out |
| 00:24.34 | *** part/#asterisk oliverqg (n=oliverqg@dsl081-096-215.den1.dsl.speakeasy.net) |
| 00:25.04 | justinu | i'm playing with dial macros to do what I want, but it's not cooperating |
| 00:25.41 | justinu | i'm sitting here looking at the source to app_macro.c |
| 00:25.54 | generalhan | how come there is no option for "mailbox size = 100" in voicemail.conf like there is for maxlength and stuff like that ? then this task would have been MUCH easier |
| 00:26.01 | *** join/#asterisk pickard (n=pickard@ip32-134-122-200.ct.co.cr) |
| 00:26.11 | Ariel_ | generalhan, it's hard coded |
| 00:26.11 | *** join/#asterisk Simon- (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb) |
| 00:26.17 | justinu | i know this doesn't help, but who the hell needs more than 100 messages in a mailbox? |
| 00:26.23 | netsurfer | marc324 - whatever test u do make sure it involves checking for echo.. those x100p cards are pure dog shit |
| 00:26.24 | Hymie | why do webpages always have pictures of people doing things on them. "Hi, we write code.. here is a picture of some random dude on the phone.. now you like us, right?" |
| 00:26.25 | generalhan | so there is no way to get it more than that ? |
| 00:26.46 | theblue | Hymie: No idea, but it pisses me off too. |
| 00:26.48 | pickard | Hi. Has anyone worked with a z-plex 10 channel bank? I am trying to get all channels to be FXS ports. |
| 00:26.48 | file | generalhan: there's the maxmsg option |
| 00:26.50 | file | maxmsg=150 |
| 00:26.56 | justinu | lol |
| 00:27.04 | fugitivo | i'm looking for a voip provider, us numbers |
| 00:27.06 | fugitivo | any recommendation? |
| 00:27.11 | justinu | broadvoice |
| 00:27.26 | generalhan | jusintu: i do. see i have a set of instructions to weed out certain people that call the Law Office that either shouldnt or that we dont want to waste time answering, so they go directly to a VM Box that gets checked daily, but my list grows daily too and 100 just isnt enough |
| 00:27.27 | marc324 | how do you send a dial cmd using asterisk and x100p card? |
| 00:27.37 | file | generalhan: see what I said. |
| 00:27.38 | Ariel_ | fugitivo, asterlink, voicepulse, |
| 00:27.43 | justinu | i see |
| 00:27.55 | fugitivo | Ariel_: sip? |
| 00:28.07 | generalhan | i dont have a "maxmsg" line in my voicemail.conf |
| 00:28.11 | xheliox | Teliax |
| 00:28.17 | Ariel_ | asterlink has sip and iax so does voicepulse |
| 00:28.20 | file | generalhan: add one. |
| 00:28.25 | generalhan | lol |
| 00:28.26 | generalhan | really |
| 00:28.31 | Ariel_ | broadvoice is sip only but there bad some times |
| 00:28.34 | file | Asterlink does not do US numbers though, besides toll-free... |
| 00:28.35 | Hymie | hmm |
| 00:28.45 | generalhan | yeai use Voicepulse, but they arent soo good for us right now |
| 00:28.48 | justinu | hmm, we've had good luck with broadvoice |
| 00:28.52 | justinu | but we're using it to call taiwan all the time |
| 00:28.53 | justinu | not US |
| 00:29.08 | justinu | they have unlimited LD to 21 countries for 20 bucks a month or something |
| 00:29.09 | toddf | voxee.com works good for me to call to US numbers |
| 00:29.20 | Ariel_ | today there was a problem with many providers due to cogent and L3 not peering |
| 00:29.21 | Hymie | instead of a timed command running in a channel.. is there a global timed event? |
| 00:29.27 | justinu | asterlink.com doesn't even work |
| 00:29.44 | justinu | oh, someone dropped a cogent router at the peering point in LA |
| 00:29.54 | generalhan | voicepulse worked REAL well for us in our old office, but now we hit these 2 savvis.net router interfaces where we are losing about 80% of our packets, and our ISP wont do anything to route around them |
| 00:30.34 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 00:30.35 | Ariel_ | well I have voicepulses connection for just an inbound did. Has been working for over 2 years not no problems. |
| 00:31.13 | generalhan | file: i tried the maxmsg=150, and it didnt work out for me |
| 00:31.17 | generalhan | still can only hold 100 |
| 00:31.24 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 00:31.25 | Ariel_ | For outbound I use nufone, voipjet and as a backup voicepulse |
| 00:31.39 | justinu | i've heard nufone sucks too |
| 00:31.42 | Ariel_ | generalhan, which version of asterisk? |
| 00:31.46 | toddf | generalhan: you reloaded asterisk right??? I've even found sometimes you must exit and re-start it .. |
| 00:31.47 | generalhan | 1.0.9 |
| 00:31.54 | Ariel_ | generalhan, hard coded |
| 00:32.19 | generalhan | lol i know i heard you last time, but file said to add the line and i will try anything, so i had to at least try |
| 00:32.25 | Hymie | hmm.. is there a way to run commands inside asterisk, from an external script? |
| 00:32.30 | Ariel_ | generalhan, head not stable |
| 00:32.34 | file | people still use 1.0.9? sillyness |
| 00:32.38 | Hymie | like a croned bash script I write, or some such? |
| 00:32.43 | netsurfer | Hymie - System() |
| 00:32.51 | marc324 | what is the diff between te411p and te410p? |
| 00:32.53 | Ariel_ | file, lots and I will not use head until it's at least 1.2.2 |
| 00:33.00 | generalhan | i dont use ZAP channels or anything like that i have no reason to switch to HEAD, when i have things working ok for me now |
| 00:33.00 | Hymie | netsurfer: I'm looking for the other way |
| 00:33.01 | Ariel_ | echo board |
| 00:33.07 | fugitivo | Ariel_: do you have the ip address of the servers? so i can test the latency |
| 00:33.08 | Hymie | netsurfer: running something from bash to effect asterisk |
| 00:33.22 | Ariel_ | fugitivo, which one |
| 00:33.24 | marc324 | ariel-- less echp? |
| 00:33.35 | fugitivo | Ariel_: asterlink or voicepulse |
| 00:33.50 | Ariel_ | fugitivo, just a sec. |
| 00:34.07 | generalhan | fugitivo: go with Voicepulse. |
| 00:34.16 | generalhan | Talk with Chris Lui he will help you out like no other |
| 00:34.17 | fugitivo | generalhan: why? |
| 00:34.21 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
| 00:34.22 | theblue | ? |
| 00:34.24 | theblue | Hi all. |
| 00:34.32 | fugitivo | generalhan: how do i talk with him? |
| 00:34.34 | justinu | [17:34] *** theblue has signed off IRC (Nick collision from services.). |
| 00:34.34 | generalhan | and Chris' lead engineer, his name is Ravi and he is the man |
| 00:34.51 | generalhan | fugitivo: you calling from the US ? |
| 00:35.16 | fugitivo | argentina |
| 00:35.30 | fugitivo | that's why i want to test the latency |
| 00:35.32 | justinu | anyone notice that tf.voipmich.com no longer works? |
| 00:35.32 | generalhan | ohh i thought you neede US numbers |
| 00:35.43 | fugitivo | yes i need |
| 00:35.47 | justinu | i used to be able to dial US toll free PSTN numbers thru them |
| 00:36.00 | Ariel_ | fugitivo, gwiaxt01.voicepulse.com |
| 00:36.12 | file | justinu: it varies |
| 00:36.27 | fugitivo | Ariel_: thanks |
| 00:36.33 | generalhan | well it depends are you going to use their wholesale minutes or voicepulse connect service ? |
| 00:36.33 | fugitivo | 160ms avg |
| 00:36.45 | file | justinu: they're working for me right now fyi |
| 00:36.47 | Ariel_ | brb |
| 00:36.53 | marc324 | How can I setup a voicemail with x100p? |
| 00:36.56 | generalhan | Ariel_: i use wgw001.voicepulse.com |
| 00:37.12 | justinu | er tries |
| 00:37.48 | fugitivo | better than broadvoice |
| 00:37.49 | toddf | anybody can get a FreeWorldDialup.com account and call US toll free PSTN numbers |
| 00:38.22 | toddf | of course, I've seen US toll free PSTN numbers that vonage, FreeWorldDialup, _and_ voxee.com wouldn't reach, but my mobile phone does .. *sigh* .. |
| 00:38.28 | justinu | fwd just redirects it to tf.voipmich.com |
| 00:38.32 | generalhan | well i need to go, i cant stay at work any longer ! lol time to go hit the bars ! |
| 00:38.37 | generalhan | talk to you all tomorrow ! |
| 00:39.07 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 00:39.41 | justinu | hmm, still doesn't work for me |
| 00:39.52 | justinu | what sip uri are you using, file? |
| 00:40.08 | justinu | i'm inviting sip:18005558355@tf.voipmich.com |
| 00:40.37 | justinu | i just get "we're sorry, your call did not go thru, please try your call again later" |
| 00:40.39 | justinu | inband |
| 00:40.49 | Ariel_ | there is a free us iax account for 800 and some pstn calling from goiax.com |
| 00:42.36 | justinu | how can they offer free pstn access? |
| 00:42.40 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 00:43.00 | toddf | http://tinyurl.com/9vbah <-- wrt the network outages mentioned here earlier |
| 00:43.24 | fugitivo | Ariel_: i want to sign up on voicepulse, but it ask me to choose a device ? |
| 00:43.28 | fugitivo | Ariel_: any idea? :) |
| 00:43.46 | fugitivo | i don't want a sipura |
| 00:44.16 | marc324 | who uses asterisk for their network? |
| 00:44.44 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
| 00:45.12 | Ariel_ | fugitivo, go to the botton of there main page and pick connections |
| 00:46.15 | justinu | voicepulse connect, is the actual link |
| 00:46.24 | fugitivo | connect.voicepulse.com |
| 00:46.25 | fugitivo | thanks! |
| 00:46.48 | justinu | wow, 11 bucks a month for inbound did |
| 00:47.01 | justinu | i can get them for 50 cents a month wholesale |
| 00:48.09 | SkramX | Of course, its WHOLESALE. But voicepulse-connect is much more viable for an end-user. |
| 00:48.12 | SkramX | :? |
| 00:48.22 | justinu | i guess, but that's still steep |
| 00:48.27 | justinu | other people are doing 8 bucks a month |
| 00:48.37 | fugitivo | anyone tried alphaphone? |
| 00:49.13 | SkramX | justinu: who? |
| 00:49.44 | justinu | broadvoice, i think |
| 00:50.00 | justinu | some other nonames |
| 00:52.46 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 00:53.39 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 00:53.42 | toddf | justinu: I've not seen cheaper than www.libretel.com for inbound did, $6/mo |
| 00:53.47 | justinu | oh yeah |
| 00:53.53 | toddf | thats what I use |
| 00:53.53 | justinu | libretel was the other one |
| 00:53.59 | justinu | but they only offer a few east coast places |
| 00:54.08 | justinu | east coast NPAs |
| 00:54.21 | SkramX | libretel reliable? |
| 00:54.24 | toddf | a friend's company is hooking up with level3, have told me they'd do two #'s for $5/mo each for me .. but they don't have my state yet .. |
| 00:54.37 | justinu | yeah, i'm hooking up with level3 also |
| 00:54.43 | toddf | SkramX they forward to FWD for me for now, I don't know how to use their sip url scheme to get it to go to my asterisk box directly |
| 00:54.44 | Ariel_ | justinu, if you just need inbound us number I think that stanaphone offers a free ny number |
| 00:54.46 | SkramX | "We are sorry, but we've currently stopped providing new individual accounts If you are interested in becoming a registered reseller, please see the reseller button on the home page" |
| 00:55.37 | SkramX | Aswell as ipkall.com (360 numbers) |
| 00:55.52 | Ariel_ | another one that is fairly cheap is sipphone they have a free number system then have addons for did's to that account. |
| 00:56.42 | SkramX | alpaphone support isnt picking up... not very good support :| |
| 00:56.42 | fugitivo | stanaphone is free |
| 00:56.57 | fugitivo | free incoming calls |
| 00:57.07 | fugitivo | new york number |
| 00:57.56 | Micc | I need a windows iax phone that actually works. |
| 00:58.15 | fugitivo | Micc: firefly (cpu bug in some machines), iaxcomm |
| 00:58.15 | *** join/#asterisk flenders (n=fserto@61.8.29.101) |
| 00:58.34 | Micc | fugitivo, iaxcomm has audio gaps for me. |
| 00:58.40 | fugitivo | firefly? |
| 00:58.47 | marc324 | any big comp using asterisk? |
| 00:58.52 | Micc | never tried firefly. Where can I get it. |
| 00:59.03 | fugitivo | google |
| 00:59.14 | Micc | does asterisk have problems on dual procs? |
| 01:00.17 | *** join/#asterisk goatmilk (n=goatmilk@130-127-45-11.chouse.resnet.clemson.edu) |
| 01:00.18 | SkramX | marc324: a bunch. |
| 01:00.44 | SkramX | Micc: No, I know a couple people who have done so on gentoo. |
| 01:00.58 | SkramX | I will be trying on dual 700mhz P3's running debian shortly |
| 01:01.49 | marc324 | is there a gui for asterisk.. i dont see myself typing in vi for a long time. |
| 01:02.24 | SkramX | marc324: something called Asterisk At Home |
| 01:02.31 | SkramX | marc324: for configurations or routing of calls? |
| 01:02.39 | SkramX | asternic.org for routing calls. |
| 01:03.21 | marc324 | config |
| 01:04.00 | SkramX | http://www.voip-info.org/wiki-Asterisk+GUI |
| 01:09.30 | Micc | firefly has problems with audio too. |
| 01:09.54 | Micc | I have an extension that just loops and plays mp3s. |
| 01:10.18 | Micc | I call with iaxcomm and it has gaps in the audio, it cuts out every once in a while. |
| 01:16.14 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 01:16.14 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 01:16.21 | Cresl1n | ~jbot |
| 01:16.22 | jbot | extra, extra, read all about it, jbot is dumb |
| 01:16.37 | Micc | fugitivo, it happens with or without a sound card. |
| 01:16.39 | Cresl1n | huh, who'd have thought? |
| 01:17.31 | Micc | fugitivo, we dump it right to an encoder and broadcast it and it has the same problem. |
| 01:17.47 | Micc | fugitivo, tried this on a number of different machines. |
| 01:18.00 | Micc | I'm kind of thinking that IAX is just broken. |
| 01:18.38 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:21.23 | *** join/#asterisk yartelecom (n=no-email@as-ferg1.yartelecom.net) |
| 01:25.05 | *** join/#asterisk MrMAGO (n=mglucksm@pdpc/supporter/sustaining/MrMAGO) |
| 01:26.19 | *** part/#asterisk Hymie (i=hymie@L8R.net) |
| 01:26.40 | MrMAGO | good night everyone... anyone knows of AstCC leaving cards in use and not charging calls? |
| 01:26.48 | Cresl1n | mmm |
| 01:38.34 | *** join/#asterisk rene- (n=root@dsl-201-144-10-211.prod-infinitum.com.mx) |
| 01:38.51 | rene- | hello all |
| 01:38.58 | SkramX | hwllo. |
| 01:40.01 | rene- | im having this very weird issue with asterisk, it wont playback audio!! calls betwwen extensions are fine, bt any call to asterisk apps like voicmail or agent login where * is supposed to send audio well it doesnt |
| 01:40.16 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 01:41.00 | SkramX | does it show anything in the CLI |
| 01:41.21 | rene- | yeah it shos somethin like playback audiofilename bu no audio at all |
| 01:42.09 | SkramX | and you are trying this via a hard ip phone? softphone? protocol? PSTN? |
| 01:42.13 | SkramX | telepathy? |
| 01:42.16 | rene- | firewall is disabled, sip.conf general section has suitable codecs, the setup worked before what could it be wrong |
| 01:42.21 | rene- | SIP |
| 01:42.39 | rene- | local subnet |
| 01:42.56 | rene- | im using uniden hardphones uip200 model |
| 01:43.06 | SkramX | Hmm. |
| 01:43.21 | SkramX | so playback() and background() both, do not work? |
| 01:43.32 | rene- | yeah, and record fails also |
| 01:43.54 | SkramX | reinstall asterisk |
| 01:43.57 | SkramX | haha, I am not too sure? |
| 01:43.59 | rene- | because ai have an extension that used to record files and even tho i a not getting the beep indication to record all files recorded are zero bytes |
| 01:44.09 | SkramX | did you go on a drunkard expedition through the C files lately? |
| 01:44.25 | rene- | haha no? |
| 01:44.26 | SkramX | :P |
| 01:44.48 | *** join/#asterisk Jameno123 (n=james@ddsl-216-68-219-38.fuse.net) |
| 01:44.53 | rene- | lets try that |
| 01:46.38 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:47.41 | SkramX | Have phun, |
| 01:49.19 | Cresl1n | no phooling around here |
| 01:53.24 | *** join/#asterisk h2oconsulting (i=H2O@ppp-69-213-7-106.dsl.kntpin.ameritech.net) |
| 01:54.15 | h2oconsulting | can anyone help guide a new guy to where he can find information about call routing |
| 01:54.28 | h2oconsulting | I am trying to route calls by incomeing trunk |
| 01:56.04 | rene- | B |
| 01:56.10 | rene- | didnt helped that either |
| 01:56.22 | SkramX | h2oconsulting: and you are going to make money off this newly acquired knowledge that someone tells you? |
| 01:56.46 | h2oconsulting | nope just useing it for my house |
| 01:56.54 | SkramX | haha |
| 01:57.06 | h2oconsulting | I do windows consulting, not good enough on linux |
| 01:57.13 | rene- | SkramX have you seen anything like it before? |
| 01:57.14 | SkramX | Okay, so you are trying to route incoming or outgoing connections |
| 01:57.20 | SkramX | rene-: like what? |
| 01:57.23 | h2oconsulting | incomeing |
| 01:57.32 | rene- | like no audio from asterisk |
| 01:57.54 | SkramX | so.. like an IVR or depending on how they call in, they get sent to a certain extension |
| 01:57.58 | SkramX | rene-: no.. |
| 01:58.26 | h2oconsulting | yeah I was hopeing for it to play a message then go to a extention |
| 01:58.41 | rene- | well ill try the olde reboot thing |
| 01:58.41 | SkramX | So... everyone goes to the same IVR though, right? |
| 01:58.50 | SkramX | have you ever used asterisk? is it installed? |
| 01:59.53 | h2oconsulting | yes I am useing asterisk@home |
| 02:00.22 | *** join/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 02:00.41 | SkramX | hello Uberbot |
| 02:01.23 | *** part/#asterisk pickard (n=pickard@ip32-134-122-200.ct.co.cr) |
| 02:01.53 | MrMAGO | anyone knows why would astcc leave a card in use and not charge? |
| 02:04.52 | Supaplex | just because |
| 02:07.56 | *** join/#asterisk marc324 (n=marc3234@206-248-159-4.dsl.teksavvy.com) |
| 02:08.27 | marc324 | is there a xwin interface for asterisk? |
| 02:10.00 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 02:10.25 | mmlj4 | if I have an extension 12345, should the voicemailbox also be 12345, or something else? (will there be any sort of conflict if the numbers are the same? |
| 02:10.57 | SkramX | nope. |
| 02:11.13 | SkramX | extensions are defined in extensions.conf, mailboxes in voicemail.conf. |
| 02:11.18 | SkramX | mmlj4: okay? |
| 02:12.20 | marc324 | why has no one ever built a gui for all the config files? |
| 02:13.09 | SkramX | marc324: go ahead, make one. |
| 02:13.37 | SkramX | is you use realtime asterisk, because of the MYSQL integration, a PHP-based GUI would be somewhat simple |
| 02:14.02 | mog_home | yeah but realtime isnt totally all through asterisk yet |
| 02:14.05 | mog_home | better to do that first |
| 02:14.28 | SkramX | Correct-o. |
| 02:14.30 | marc324 | sounds like a alpha software |
| 02:15.03 | *** join/#asterisk flenders (n=fserto@61.8.29.101) |
| 02:15.33 | mog_home | whats alpha software? |
| 02:16.45 | marc324 | this app looks like a beast to maintain. |
| 02:17.03 | mog_home | asterisk? |
| 02:17.14 | mog_home | takes less than an hour to do mild config |
| 02:17.21 | mog_home | its no worse than apache |
| 02:17.27 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 02:17.28 | theblue | ? |
| 02:17.39 | mmlj4 | SkramX: thanks |
| 02:17.59 | SkramX | sure.. |
| 02:18.32 | SkramX | marc324: we can help you.. |
| 02:19.35 | marc324 | most of the config files can be translated into nice gui, I do not understand why one has to go and edit the files. |
| 02:20.07 | mog_home | its not nearly as bad as you make it sound |
| 02:20.13 | mog_home | guis take a way options |
| 02:20.17 | mog_home | or look ugly as ass |
| 02:20.39 | mtgh | marc324: Feel free to write one |
| 02:21.21 | SkramX | marc324: pay me and ill make you a full GUI. |
| 02:21.22 | SkramX | hehe |
| 02:21.26 | *** join/#asterisk juice (n=juice@mo-67-77-177-133.dyn.sprint-hsd.net) |
| 02:21.44 | mog_home | ew |
| 02:21.45 | *** join/#asterisk trig_hm (n=jb@home.monkeypr0n.org) |
| 02:22.46 | marc324 | a web based control panel. |
| 02:22.54 | marc324 | 2005 |
| 02:23.15 | mog_home | ? |
| 02:25.04 | *** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr) |
| 02:25.28 | jskcr | hy all |
| 02:25.32 | mog_home | hi |
| 02:25.35 | SkramX | Hello. |
| 02:25.38 | file | owwwwwwwww |
| 02:26.31 | *** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 02:26.47 | jskcr | Hya SkramX, whats happening |
| 02:27.11 | JerJer | mmmm yeah |
| 02:27.32 | JerJer | i'm gonna go ahead and need you to come in this saturday |
| 02:27.44 | JerJer | we've lost a few people and have to play catch up |
| 02:28.03 | JerJer | oh and i'm gonna also need you to come in on Sunday to....mkay |
| 02:28.10 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 02:28.26 | JerJer | thanks |
| 02:28.31 | mog_home | lol jerjer |
| 02:28.47 | SkramX | jskcr: not too much. |
| 02:29.11 | jskcr | SkramX: Im busy building a 64bit intel gentoo box :) |
| 02:29.13 | SkramX | Veto_laptop: San Antonio, aye? |
| 02:29.33 | SkramX | jskcr: fun. Just got a celeron 2.4ghz dedi at theplanet.com |
| 02:29.55 | jskcr | SkramX: the d915gag I believe digium had some tdm problems with em |
| 02:30.09 | SkramX | i dont do hardware. |
| 02:30.20 | Cresl1n | with em? |
| 02:30.25 | Veto_laptop | SkramX: Close, New Braunfels |
| 02:30.26 | Cresl1n | how's that? |
| 02:30.27 | jskcr | SkramX: its a nice cheap intel desktop board |
| 02:30.36 | JerJer | Cresl1n: you mean nose and throat? |
| 02:30.38 | SkramX | Veto_laptop: fun, fun.. I am in the ATX (Austin, TX) |
| 02:30.49 | Cresl1n | lol |
| 02:30.54 | Cresl1n | too much fn |
| 02:30.57 | mog_home | new braunfles thats where my grandparents used to live |
| 02:31.03 | Cresl1n | JerJer: are you going to astricon? |
| 02:31.08 | mog_home | we used to go to shliderbaun |
| 02:31.18 | mog_home | was loads of fun |
| 02:31.22 | JerJer | Cresl1n: yep, i was properly motivated to go |
| 02:31.28 | Cresl1n | sweet! |
| 02:31.36 | Cresl1n | who motivated you? |
| 02:31.54 | *** part/#asterisk h2oconsulting (i=H2O@ppp-69-213-7-106.dsl.kntpin.ameritech.net) |
| 02:32.01 | jskcr | Cresl1n: http://www.digium.com/index.php?menu=compatibility its the d915gag on the bottom |
| 02:32.08 | jskcr | it seems to work great now with em |
| 02:32.28 | SkramX | mog_home: where do you live now? |
| 02:32.38 | SkramX | Veto_laptop: grew up in Texas or what |
| 02:32.39 | mog_home | well they live in mobile, al now with my folkes |
| 02:32.43 | JerJer | Cresl1n: out of the blue I had a very early customer just pop back online and ask if i was attending. I told him no, due to financial reasons, so he proceeded to cash in some frequent flier miles to get my ass out there for the week |
| 02:32.45 | mog_home | i live in huntsville with digium |
| 02:32.51 | mog_home | my brother is out in waco though |
| 02:33.00 | mog_home | nice |
| 02:33.10 | Cresl1n | JerJer: that's pretty nice |
| 02:33.27 | jskcr | JerJer wow thats cool |
| 02:33.38 | jskcr | I have to work :( |
| 02:33.49 | JerJer | yeah i thought so too. He says my company needs to be out there, so now we are. |
| 02:34.00 | Cresl1n | :-D |
| 02:34.20 | JerJer | i am going to attempt to launch our new website |
| 02:34.36 | SkramX | mog_home: fun. |
| 02:34.46 | Cresl1n | :-D |
| 02:34.57 | *** join/#asterisk MrMAGO (n=mglucksm@pdpc/supporter/sustaining/MrMAGO) |
| 02:34.58 | JerJer | only worry is not blowing up a stupid amount of DIDs |
| 02:35.01 | mog_home | whats that me cresl1n |
| 02:35.19 | SkramX | :| |
| 02:35.22 | JerJer | lol |
| 02:35.23 | file[laptop] | you two are rebels |
| 02:35.31 | SkramX | crazy kids |
| 02:35.43 | mog_home | open source telephony does strange things to ya |
| 02:35.48 | Cresl1n | yeah |
| 02:35.55 | Cresl1n | stupid telephony |
| 02:35.58 | Cresl1n | worse than the cold |
| 02:36.09 | mog_home | its like herpes |
| 02:36.12 | mog_home | dissapears sometimes |
| 02:36.15 | Cresl1n | ewwww |
| 02:36.16 | mog_home | but you always have it |
| 02:36.21 | mog_home | sometimes you infect some one else |
| 02:36.22 | file[laptop] | mog_home: yeah like no... |
| 02:36.27 | Cresl1n | that's the sickest example I've ever heard |
| 02:36.29 | mog_home | lol |
| 02:36.32 | file[laptop] | if I was a telephony guy, I'd have all the phones in the world! |
| 02:37.04 | mog_home | my number is perputually out of service |
| 02:37.22 | hypa7ia | in the PReye? |
| 02:37.33 | file[laptop] | "We're sorry, the number you are attempting to dial is currently going to a country that no longer exists. Please go back in time and try your call. Thank you." |
| 02:37.46 | Cresl1n | did allison say that? |
| 02:37.47 | mog_home | heh |
| 02:37.47 | Cresl1n | :-D |
| 02:37.58 | file[laptop] | no but I'm tempted to put it on our next recording list |
| 02:37.58 | mog_home | my parents think its so funny my number never works |
| 02:38.21 | Cresl1n | cell phone.... |
| 02:38.33 | mog_home | lol never |
| 02:38.37 | mog_home | never never never |
| 02:38.43 | mog_home | unless bill buys me one |
| 02:38.58 | SkramX | do most of yallw ork for digium/ |
| 02:39.04 | Cresl1n | no |
| 02:39.05 | mog_home | well cresl1n and me |
| 02:39.05 | Cresl1n | just two of us |
| 02:39.05 | SkramX | oh, file[laptop] works for asterlink... |
| 02:39.07 | Cresl1n | :-) |
| 02:39.10 | SkramX | Cresl1n: and mog_home |
| 02:39.11 | SkramX | o ok |
| 02:39.24 | mog_home | we are teh digium 1337 haxors |
| 02:39.27 | SkramX | how many work for digium in all? 10? |
| 02:39.30 | mog_home | 50 |
| 02:39.33 | Cresl1n | like 40 |
| 02:39.35 | Cresl1n | 50? |
| 02:39.37 | mog_home | yeah |
| 02:39.38 | Cresl1n | not yet |
| 02:39.38 | file[laptop] | 45! |
| 02:39.41 | mog_home | you need to keep up |
| 02:39.50 | mog_home | 48 plus the two new people |
| 02:39.55 | mog_home | that come on next week |
| 02:39.56 | SkramX | all full time? |
| 02:39.57 | Cresl1n | no way! |
| 02:39.58 | file[laptop] | in other news, 21 minutes till I'm 19 |
| 02:39.59 | mog_home | or the next one |
| 02:40.02 | mog_home | yeah |
| 02:40.03 | SkramX | file[laptop]: fun. |
| 02:40.07 | SkramX | College? |
| 02:40.11 | Cresl1n | no, a significant chunk are part time |
| 02:40.14 | SkramX | or just working |
| 02:40.15 | mog_home | yeah |
| 02:40.16 | hypa7ia | file[laptop]: you could drink in ontario! |
| 02:40.19 | file[laptop] | SkramX: not yet |
| 02:40.22 | mog_home | over half are part or hourly |
| 02:40.24 | SkramX | A while until I am in college. |
| 02:40.29 | mog_home | like i am hourly and in college |
| 02:40.29 | file[laptop] | hypa7ia: and here in NB |
| 02:40.33 | mog_home | but hit 45 hours most weeks |
| 02:40.36 | SkramX | file[laptop]: high school or just hanging and doing the voip-work thang. |
| 02:40.48 | file[laptop] | I graduated high school in June... I think it was... |
| 02:40.52 | SkramX | hehe |
| 02:40.53 | Cresl1n | pfff |
| 02:40.58 | Cresl1n | it wasn't that long ago file |
| 02:40.59 | Cresl1n | :-D |
| 02:41.02 | file[laptop] | and transferred to full time in August... or June... meh I forget |
| 02:41.10 | hypa7ia | file[laptop]: oh yea didn;t know you were up here in canuckistan |
| 02:41.24 | file[laptop] | ah July |
| 02:41.31 | file[laptop] | hypa7ia: yes I am in Atlantic Canada eh |
| 02:41.45 | hypa7ia | cool |
| 02:42.05 | Cresl1n | what's a vouch? :-D |
| 02:42.14 | file[laptop] | I could go for a Tim Horton's donut right now |
| 02:42.25 | file[laptop] | Cresl1n: Freestyler! |
| 02:42.38 | Cresl1n | file: oh no you didn't! |
| 02:43.03 | file[laptop] | Cresl1n: SIP it! |
| 02:43.09 | SkramX | hahaha |
| 02:43.11 | Cresl1n | SIP happnes |
| 02:43.19 | file[laptop] | all too often |
| 02:43.25 | *** join/#asterisk jaike (n=a@203.177.36.132) |
| 02:44.15 | *** join/#asterisk ZX81 (n=ZX81@222-153-100-242.jetstream.xtra.co.nz) |
| 02:44.16 | ZX81 | hi all |
| 02:44.22 | ZX81 | how do I get rid of Oct 5 22:45:22 WARNING[3075]: chan_sip.c:3319 process_sdp: Unknown SDP media type in offer: image 31914 udptl t38 |
| 02:44.25 | file[laptop] | eep it's Matt |
| 02:44.29 | ZX81 | :) |
| 02:44.33 | file[laptop] | ZX81: disable T.38? |
| 02:44.35 | ZX81 | heelo |
| 02:44.38 | ZX81 | grrr can't |
| 02:44.47 | file[laptop] | modify the source? disable warnings? |
| 02:44.49 | ZX81 | long voip path |
| 02:44.50 | SkramX | ? |
| 02:44.54 | ZX81 | yeah done |
| 02:44.57 | ZX81 | but cpu goes up |
| 02:44.59 | ZX81 | :) |
| 02:45.03 | ZX81 | maybe my imagination |
| 02:45.05 | ZX81 | one sec |
| 02:45.09 | file[laptop] | probably your imagination |
| 02:45.34 | file[laptop] | ZX81: I upgraded the MySQL on quark btw, was going to give you notice but you didn't answer :P |
| 02:46.42 | ZX81 | :D |
| 02:46.44 | ZX81 | kewl |
| 02:46.46 | ZX81 | still works? |
| 02:46.50 | file[laptop] | yes |
| 02:46.54 | ZX81 | yep |
| 02:46.55 | ZX81 | :) |
| 02:46.57 | file[laptop] | I only broke client stuff, but I fixed that |
| 02:47.02 | ZX81 | first coffee just ready now |
| 02:47.04 | ZX81 | :) |
| 02:47.10 | file[laptop] | haha |
| 02:47.13 | *** join/#asterisk protien (i=jjmtrev@203-173-26-187.dyn.iinet.net.au) |
| 02:47.27 | ZX81 | I have a client doing 22kminutes with like 10% T.38 |
| 02:47.37 | ZX81 | and the T.38 takes out the console something cronic |
| 02:47.38 | ZX81 | :) |
| 02:47.59 | ZX81 | when I changed logged.conf so that warning weren't displayed |
| 02:48.02 | file[laptop] | coppice has some passthrough stuff that supposedly may work... I dunno, I don't have any T38 stuff |
| 02:48.04 | ZX81 | the cpu went to 49% |
| 02:48.08 | ZX81 | yeah |
| 02:48.10 | ZX81 | I knwo |
| 02:48.14 | ZX81 | I commented on it |
| 02:48.24 | ZX81 | maybe I should just try it out on production :D |
| 02:48.34 | file[laptop] | great idea!!! *G* |
| 02:48.38 | ZX81 | :D |
| 02:48.48 | ZX81 | brb |
| 02:49.02 | file[laptop] | okay folks - nothing to see here, move along... |
| 02:49.21 | JunK-Y | moooo |
| 02:50.02 | file[laptop] | Junky!!! |
| 02:50.12 | JunK-Y | file!!!! |
| 02:50.27 | file[laptop] | how are you? |
| 02:50.35 | JunK-Y | im fine urself? |
| 02:50.48 | file[laptop] | not too bad |
| 02:51.13 | *** join/#asterisk neil_ablang (n=neil@202.124.128.39) |
| 02:51.54 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 02:52.12 | neil_ablang | any tips of making e&m_w work with dms300? i got a te411p card |
| 02:52.54 | neil_ablang | was able to send calls, but receiving calls seems not working |
| 02:56.44 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 02:57.55 | *** join/#asterisk hacim (i=micah@debian/developer/micah) |
| 02:58.09 | hacim | anyone know what format the broadvoice config files are in? its apparantly binary |
| 02:59.45 | blitzrage | this is just too damn cool: http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-the-future-of-telephony-book.asp |
| 02:59.51 | blitzrage | I've been quoted! |
| 03:00.11 | file[laptop] | :) |
| 03:00.12 | Corydon76-home | woohoo! |
| 03:00.21 | hypa7ia | blitzrage: nice! |
| 03:00.53 | blitzrage | hypa7ia: oh yah - I'm smiling large :) |
| 03:01.23 | file[laptop] | yay blitzrage |
| 03:02.22 | Pete_Largo | very nice quote bliztrage :) |
| 03:02.32 | blitzrage | Pete_Largo: thanks :) you know which author I am? :) |
| 03:02.41 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 03:02.45 | Pete_Largo | I believe so |
| 03:02.57 | blitzrage | Pete_Largo: lol |
| 03:03.12 | mmlj4 | what's could cause voicemailmain to not see keypresses? -- No username but # key pressed. Using CID '2076' / -- Playing 'vm-password' (language 'en') / -- Incorrect password '' for user '2076' (context = <any>) |
| 03:03.34 | Pete_Largo | rhymes with leaf |
| 03:03.39 | Pete_Largo | ;) |
| 03:03.40 | mmlj4 | teef? |
| 03:03.48 | file[laptop] | doesn't rhyme with leaf |
| 03:04.00 | Pete_Largo | it was a joke file |
| 03:04.01 | Pete_Largo | haha |
| 03:04.07 | Sedorox | dtmf |
| 03:04.18 | file[laptop] | LIFE! |
| 03:04.22 | blitzrage | yes? :) |
| 03:04.31 | file[laptop] | no, you're Leif |
| 03:04.34 | Pete_Largo | "Asterisk is arguably the most influential and exciting piece of software since the operating system it runs on--Linux," |
| 03:04.36 | file[laptop] | there's a difference. |
| 03:05.02 | Corydon76-home | Rhymes with queef |
| 03:05.13 | Pete_Largo | then I am pronouncing it wrong. just like I pronounce Linux wrong |
| 03:05.14 | blitzrage | Corydon76-home: then you don't know how to say my name :) |
| 03:05.35 | blitzrage | <--- Leif == Life |
| 03:05.51 | Corydon76-home | Asstricks is another channel... |
| 03:05.57 | Pete_Largo | I think you say that phrase a lot George :) |
| 03:05.59 | FuriousGeorge | then i discovered theres another syllable |
| 03:06.07 | blitzrage | FuriousGeorge: I always try to over pronounciate "Asterisk" |
| 03:06.14 | blitzrage | FuriousGeorge: exactly :) |
| 03:06.20 | file[laptop] | blitzrage: guess what book arrives later today! |
| 03:06.24 | Corydon76-home | And don't forget, it's nucular |
| 03:06.24 | blitzrage | file[laptop]: w00t! |
| 03:06.33 | Pete_Largo | lmao nucular |
| 03:06.40 | mmlj4 | computers for dummies? # /me runs |
| 03:06.50 | file[laptop] | rm -rf mmlj4 |
| 03:06.54 | mmlj4 | heh |
| 03:06.55 | FuriousGeorge | confortable is another one |
| 03:07.00 | blitzrage | hypa7ia: birds and bees mostly |
| 03:07.06 | FuriousGeorge | people wanna say "conftable" around these parts |
| 03:07.06 | hypa7ia | alot is the worst |
| 03:07.13 | hypa7ia | ALOT IS NOT A WORD PEOPLE |
| 03:07.14 | blitzrage | a lot is TWO words |
| 03:07.20 | blitzrage | </rage> |
| 03:07.21 | hypa7ia | blitzrage: EYS |
| 03:07.22 | hypa7ia | err |
| 03:07.24 | hypa7ia | YES |
| 03:07.26 | FuriousGeorge | i hate when people mispel alot |
| 03:07.31 | file[laptop] | it's my birthday people! |
| 03:07.38 | blitzrage | Happy Birthday file |
| 03:07.40 | mmlj4 | congrats |
| 03:07.40 | Sedorox | happy b-day |
| 03:07.41 | file[laptop] | thx |
| 03:07.42 | Nivex | file[laptop]: Hippo Birdie, two ewes! |
| 03:07.44 | Pete_Largo | happy b-day |
| 03:07.45 | FuriousGeorge | cumpleanos felizes |
| 03:07.45 | *** join/#asterisk epoch (n=epoch@octane.breakbeats.org) |
| 03:07.49 | Corydon76-home | Oooo, 19... |
| 03:07.50 | hypa7ia | hey file[laptop] |
| 03:07.59 | hypa7ia | do you know the_p0pe or msvi? |
| 03:08.03 | *** part/#asterisk epoch (n=epoch@octane.breakbeats.org) |
| 03:08.23 | blitzrage | file[laptop] can drink now |
| 03:08.27 | blitzrage | (legally) |
| 03:08.27 | Corydon76-home | Once again, in your prime... |
| 03:08.32 | file[laptop] | yes, yes I can |
| 03:08.43 | Corydon76-home | That won't happen again for another 4 years |
| 03:09.30 | file[laptop] | pesky buggers |
| 03:09.36 | blitzrage | ok, I gotta go and pack for tomorrow -- big day of heading to California! |
| 03:09.44 | file[laptop] | have fun blitzrage |
| 03:09.46 | blitzrage | (tomorrow is going to be a long day...) |
| 03:10.10 | Corydon76-home | Hah... |
| 03:10.24 | Corydon76-home | You don't know what a long day is until you've been to Phreaknic... :-P |
| 03:10.37 | blitzrage | Corydon76-home: oh I know a long day... :) |
| 03:10.56 | Corydon76-home | 8 am to 4 am? |
| 03:11.01 | *** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 03:11.13 | Corydon76-home | Lather, rinse, repeat |
| 03:11.22 | jdv79 | is trunk solid:) i just tried to build it and i got some error mentioning curl something |
| 03:11.25 | glm2k | blitzrage: i'll welcome you at the border >:) |
| 03:11.34 | blitzrage | glm2k: cool :) |
| 03:11.39 | Pete_Largo | you need to have curl installed jdv79 |
| 03:11.48 | blitzrage | Corydon76-home: I was up till 5am last night |
| 03:11.53 | Corydon76-home | Not curl. libcurl |
| 03:12.00 | *** join/#asterisk JunK-Y (n=junky@69.156.123.108) |
| 03:12.11 | Corydon76-home | Same codebase, two different packages |
| 03:12.21 | Pete_Largo | my bad |
| 03:12.30 | Pete_Largo | I thought you needed both |
| 03:12.34 | Corydon76-home | The app, not the library |
| 03:12.46 | Corydon76-home | Nope, the library is enough to get the app running |
| 03:13.05 | Pete_Largo | well, I just learned something new :) |
| 03:13.09 | *** join/#asterisk Jzalae (n=sk@216-220-249-56.midmaine.com) |
| 03:13.16 | Corydon76-home | but if you're running HEAD, it's really no longer an app... it's now a function... |
| 03:13.54 | file[laptop] | Corydon76-home: HA |
| 03:13.57 | Pete_Largo | \get a room! |
| 03:13.58 | blitzrage | functions rock |
| 03:14.16 | Corydon76-home | Yep, I'm glad I helped architect functions |
| 03:14.49 | Corydon76-home | #2278 |
| 03:15.16 | Corydon76-home | Still waiting on #2720, though |
| 03:16.27 | blitzrage | Corydon76-home: what was 2720 again? |
| 03:16.37 | Corydon76-home | Stack apps |
| 03:16.47 | blitzrage | Corydon76-home: someone should really make ${DBdel()} |
| 03:16.50 | Corydon76-home | Gosub/Return/Pop, and LOCAL variables |
| 03:17.00 | blitzrage | ahh |
| 03:17.22 | Corydon76-home | blitzrage: Set(DB(foo/bar)=) |
| 03:17.56 | blitzrage | Corydon76-home: that doesn't delete the family / key though does it? |
| 03:18.03 | Corydon76-home | Nope |
| 03:18.10 | jdv79 | i did install curl |
| 03:18.13 | jdv79 | it might be out of date though |
| 03:18.23 | Corydon76-home | Install the libcurl-devel package |
| 03:18.54 | jdv79 | ah |
| 03:21.24 | *** part/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 03:21.55 | Corydon76-home | blitzrage: the DBdel and DBdeltree apps were never deprecated |
| 03:22.11 | Corydon76-home | So they'll still be around |
| 03:25.43 | *** part/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
| 03:26.21 | Pete_Largo | does the O'Reilly book talk about all the changes in 1.2 beta versus 1.0.x? |
| 03:28.17 | JonR800 | it said it covers 1.2.. how much i don't know |
| 03:29.33 | *** part/#asterisk OzJames79 (n=opera@203.208.64.29) |
| 03:29.47 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 03:32.32 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 03:35.11 | FuriousGeorge | naiomi from rome is pretty cute |
| 03:35.45 | FuriousGeorge | something really fly about the way she just said "my calendar is correct, if you'd like to have me tonight" |
| 03:37.14 | *** join/#asterisk hhrp (i=zloydyad@c-66-176-186-242.hsd1.fl.comcast.net) |
| 03:37.16 | *** part/#asterisk hhrp (i=zloydyad@c-66-176-186-242.hsd1.fl.comcast.net) |
| 03:37.19 | *** join/#asterisk hhrp (i=zloydyad@c-66-176-186-242.hsd1.fl.comcast.net) |
| 03:37.23 | hhrp | hi |
| 03:37.42 | hhrp | could anyone tell why am I having this warning and no audio when phone is ringing Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) |
| 03:39.42 | Pete_Largo | if anyone cares, I'm still looking for a sample config file for a Mediatrix 2102... |
| 03:41.45 | *** join/#asterisk dca (n=dca@c-24-8-125-67.hsd1.co.comcast.net) |
| 03:43.42 | *** join/#asterisk spackle (n=spackle@209.234.83.19) |
| 03:44.20 | *** join/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 03:44.25 | *** part/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 03:44.36 | *** join/#asterisk BrianR___ (i=brianr@c-24-61-206-174.hsd1.ma.comcast.net) |
| 03:45.20 | hhrp | could anyone tell why am I having this warning and no audio when phone is ringing Asked to transmit frame type 256, while native formats is 2 (read/write = 2/2) |
| 03:47.05 | dca | codec error |
| 03:47.17 | dca | or more specifically mismatch |
| 03:48.09 | hhrp | i cant figure out |
| 03:48.11 | Pete_Largo | what would cause it? |
| 03:48.19 | hhrp | i know its codecs |
| 03:48.21 | Pete_Largo | cause I had something similar happen to me yesterday |
| 03:48.32 | hhrp | i have gsm on one end and g729 on another |
| 03:48.44 | hhrp | when call comes in to gsm its ok |
| 03:49.00 | hhrp | then once g729 is supposed to give rining it starts |
| 03:49.05 | hhrp | and i hear no ring |
| 03:49.23 | CoaxD | Cygwin scares me. |
| 03:49.37 | dca | k, i have to ask, you do have g729 licenses on your asterisk box right? |
| 03:49.44 | hhrp | yes |
| 03:49.50 | hhrp | its been working ok |
| 03:50.33 | *** join/#asterisk optickal (n=optikal@69.182.42.65) |
| 03:50.56 | *** join/#asterisk bmg505 (n=leon@rndf-146-11-95.telkomadsl.co.za) |
| 03:51.17 | *** join/#asterisk pashah (n=pashah@ns.itconnection.ru) |
| 03:51.19 | pashah | hello |
| 03:51.59 | dca | hhrp: what does "show g729" say? |
| 03:52.58 | *** part/#asterisk spackle (n=spackle@209.234.83.19) |
| 03:53.19 | hhrp | doesnt say anything |
| 03:53.32 | hhrp | show audio codecs does |
| 03:53.51 | pashah | having wierd problems: 1 * box, 1 te205p and 3 TDM04b, after I do modprobe wct4xxp, I get ZT_CHANCONFIG failed on channel 63: No such device or address (6), if I comment out TDM string is zaptel.conf - all fine, any ideas? |
| 03:54.00 | dca | if "show g729" doesn't say "X/Y codecs available" then you don't have an g729 codecs to use |
| 03:54.17 | hhrp | well i do |
| 03:54.31 | hhrp | and it works when recepient picks up the phone |
| 03:54.31 | dca | show g729 |
| 03:54.31 | dca | 0/0 encoders/decoders of 20 licensed channels are currently in use |
| 03:54.37 | hhrp | hmm |
| 03:54.37 | dca | you should get something like that |
| 03:54.42 | hhrp | strange |
| 03:54.57 | hhrp | they paid 20 for two lines |
| 03:55.18 | dca | and activated the key on the box? |
| 03:55.33 | dca | and put codec_g729a.so in /var/lib/asterisk/modules ? |
| 03:55.45 | hhrp | i dunno i didnt set it up to be honest |
| 03:55.55 | hhrp | yes i have that so in the libs |
| 03:55.56 | dca | might wanna ask 'em |
| 03:56.08 | hhrp | its been working ok.. |
| 03:56.25 | hhrp | then outta the sudden started doing it |
| 03:57.49 | hhrp | gotta reboot |
| 03:57.53 | hhrp | you think bad codec? |
| 03:58.09 | hhrp | i know in oh323 i can set frame= |
| 03:58.11 | *** join/#asterisk brookshire (n=shep@gateway.digium.com) |
| 03:58.20 | hhrp | but it looks like its a sip |
| 03:58.27 | file[laptop] | Mattttttttttttt |
| 03:58.30 | dca | not running screen are you? |
| 03:58.40 | hhrp | screen? |
| 03:58.42 | dca | nm |
| 03:58.44 | dca | what os? |
| 03:58.49 | hhrp | rd linux |
| 03:59.24 | Nugget | Linux is poo. |
| 03:59.39 | *** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
| 04:01.15 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 04:02.01 | distortion | is there an easy way to debug sip messages? |
| 04:02.21 | distortion | other than debug sip which doesnt say shit |
| 04:02.33 | file[laptop] | you mean sip debug? |
| 04:02.52 | Pete_Largo | increase your verbosity? |
| 04:03.16 | distortion | well maybe that is the issue, i run in -vvvgc and sip debug lists too much info |
| 04:03.29 | file[laptop] | you can sip debug a specific peer or IP |
| 04:03.36 | Pete_Largo | you can? |
| 04:03.38 | file[laptop] | sip debug peer, sip debug ip |
| 04:03.50 | FuriousGeorge | CoaxD: why does cygwin scare you |
| 04:03.55 | distortion | even then, my 7960 phone sends the invite like 12 times |
| 04:03.57 | Pete_Largo | wow, that's another thing I learned today! |
| 04:04.45 | Pete_Largo | I assume you/I can do the same with IAX? |
| 04:04.48 | distortion | is there like a cdr i should be looking at with specific reasons why the call wont go through? |
| 04:05.08 | distortion | or a more simple debug that lists why something is failing? |
| 04:06.48 | *** join/#asterisk Connor (n=billy@198-144-165-65.knx.tn.nxs.net) |
| 04:07.05 | pashah | cheers |
| 04:07.16 | *** part/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net) |
| 04:11.10 | *** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc) |
| 04:12.48 | *** part/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 04:14.09 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 04:14.40 | redder86 | hi all |
| 04:15.13 | redder86 | any recommendations on a good VoIP provider servicing Mexico call destinations? |
| 04:15.57 | redder86 | dca: what's the per-minute rate to Mexico? |
| 04:16.09 | dca | http://www.teliax.com/rates.html |
| 04:16.13 | CoaxD | You know, i wonder.. Wouldnt it be beneficial to go with a voip provider that is LOCAL to mexico? :) |
| 04:16.29 | Primer | anyone got a skinny phone working with asterisk? A friend of mine has a cisco 7920, and we're trying to connect it to my asterisk |
| 04:16.30 | CoaxD | (Okay, okay, i know that most of mexico has an ip network that is scarcely better than a 56k modem.) |
| 04:16.35 | Primer | CoaxD: fag |
| 04:16.41 | CoaxD | Primer: blah |
| 04:16.45 | redder86 | CoaxD: Telmex charges an arm and a leg for most of its services |
| 04:17.00 | Primer | I thought voip was illegal in Mexixo |
| 04:17.09 | CoaxD | Primer: hell, it probably is |
| 04:17.23 | dca | it's a "grey" area |
| 04:17.29 | *** join/#asterisk AaronP (n=Aaron@c-24-6-133-255.hsd1.ca.comcast.net) |
| 04:17.53 | redder86 | Well, 3.5 cents per minute beats NuFone's 10. |
| 04:18.07 | redder86 | dca: do they allow ulaw? |
| 04:18.15 | dca | yeah |
| 04:18.20 | dca | and g729 |
| 04:18.21 | redder86 | quality good? |
| 04:18.29 | dca | yep |
| 04:18.37 | redder86 | you have an ownership interest? |
| 04:18.54 | dca | friendly interest :) |
| 04:19.13 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 04:20.22 | redder86 | $10 initial deposit and then just keep a positive balance from there? Pay on-line with a credit card, etc? |
| 04:20.50 | dca | yeah |
| 04:20.57 | redder86 | not bad |
| 04:20.59 | *** join/#asterisk huslage (n=huslage@68.178.18.183) |
| 04:21.01 | dca | 30 day money back, etc. |
| 04:21.29 | redder86 | $10 isn't worth the hassle of getting money back for a one-time deal |
| 04:21.29 | *** join/#asterisk ManxPower (n=eric@adsl-67-65-233-194.dsl.lgvwtx.swbell.net) |
| 04:21.52 | redder86 | well, I guess if one makes minimum wage, then maybe |
| 04:22.59 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 04:23.25 | dca | hehe |
| 04:23.34 | ManxPower | ~docs |
| 04:23.35 | 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 |
| 04:23.36 | ManxPower | ~mailinglist |
| 04:23.37 | jbot | methinks 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/ |
| 04:27.05 | ManxPower | These Wifi guys are funny. They go on and on about the advantages of Wifi, but don't seem to think the fact that a microwave oven can impact your network performance is a real issue. |
| 04:27.35 | brookshire | heh.. it doesn't really |
| 04:27.49 | brookshire | but those stupid 2.4ghz cordless phones do |
| 04:28.43 | brookshire | microwave ovens are mostly shielded anyways :) |
| 04:29.47 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 04:29.53 | ManxPower | "Stupid cordless phones" |
| 04:30.20 | ManxPower | Cordless phones are one of the cornerstones of modern civilization |
| 04:30.26 | Primer | anyone got a skinny phone working with asterisk? A friend of mine has a cisco 7920, and we're trying to connect it to my asterisk. Apparently this requires a tftp server? seems rather odd... |
| 04:30.27 | *** join/#asterisk Jenna (n=cherryRe@209.8.233.161) |
| 04:31.11 | ManxPower | Primer, Skinny/SCCP is VERY much tied to CallManager. The fact that anyone has gotten them to work with Asterisk is a miricle. |
| 04:31.42 | Primer | ManxPower: this friend of mine is actually a cisco engineer who works on the 7920/7960 |
| 04:31.46 | ManxPower | brookshire, I have to replace all 3 of my cordless phones now that I have a Wifi uplink for my internet conneciton |
| 04:31.59 | ManxPower | And I don't buy the cheap ones. |
| 04:32.02 | brookshire | hehe.. |
| 04:32.11 | Primer | and I'm wondering "who the fuck designed such a ridiculous scheme for connecting a phone?" |
| 04:32.11 | brookshire | i want to outlaw them at the office |
| 04:32.29 | Primer | he claims it wasn't him |
| 04:32.32 | ManxPower | I'm sometimes on the phone for 6 hours in a day, I need a good cordless phone and a good headset |
| 04:32.54 | ManxPower | brookshire, We are thinking of encouraging them in the office. Would totally destroy the rogue Wifi APs that are poping up. |
| 04:33.14 | brookshire | boo! |
| 04:33.22 | brookshire | open wifi #1 |
| 04:33.36 | ManxPower | Primer, Cisco bought the company that originally made the SCCP/Skinny phones. |
| 04:33.40 | Primer | ManxPower: so you have skinny phones working with asterisk? |
| 04:33.49 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 04:33.50 | Primer | ManxPower: yeah, they're buying everything |
| 04:33.51 | ManxPower | Primer, No, not even I like that much pain. |
| 04:34.20 | Primer | My friend tells me they bought they bought sipura's "technology" twice, via 2 separate companies |
| 04:34.27 | Primer | err |
| 04:34.30 | ManxPower | Primer, I have many more interesting things to do than try to foce a phone to work with Asterisk. |
| 04:34.31 | Primer | s/they bought/ |
| 04:34.34 | ManxPower | Primer, He is correct. |
| 04:34.46 | Jenna | Hi! Can Asterisk be used as a commercial VOIP gateway ? I know its OS but are people/companies using it to provide VOIP solutions ? |
| 04:34.50 | iCEBrkr | Primer: What are you bitching about now? :P |
| 04:34.55 | ManxPower | Primer, Actually 3 times, if you count the Linksys licensing deal. |
| 04:34.56 | theblue | If I set an extension in sip.conf for a register => line, would calls coming in from that number automatically go to that extension? |
| 04:35.00 | Primer | ManxPower: so you're telling me this is a hopeless cause? |
| 04:37.53 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 04:37.53 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 04:37.56 | glm2k | Oyster Sauce, mmmmmmm |
| 04:38.06 | Jenna | so I was wondering if I can start an VOIP business around it. are people already doing this |
| 04:38.18 | theblue | Jenna: What do you mean, a VoIP business? |
| 04:38.41 | ManxPower | Jenna, There are many companies that use Asterisk to provide VoIP services. |
| 04:38.42 | glm2k | Jenna: asterisk is only one part of a voip business. |
| 04:38.52 | Primer | that market is already saturated too |
| 04:38.58 | theblue | And how. |
| 04:39.00 | ManxPower | But you already knew that, didn't you? Since you are on the Asterisk-Biz mailiging list. |
| 04:39.02 | iCEBrkr | Primer: That's an understatement |
| 04:39.03 | glm2k | Primer: which? |
| 04:39.31 | ManxPower | Primer, Inbound Toll Free and Out bound Toll is saturated. I think there's still a market for inbound DID |
| 04:39.31 | Primer | the voip market |
| 04:40.05 | Jenna | Primer: saturated. hmm. I wonder if I can get teeny tiny bit of the existing market slice |
| 04:40.11 | ManxPower | Everyone and their brother is offering outbound calling. |
| 04:40.14 | Primer | apparently my friend is the engineer responsible for the sccp implementation on the cisco 7920 |
| 04:40.16 | glm2k | ManxPower: there are also niche markets - areas with pretty steep per minute charges |
| 04:40.17 | theblue | There IS no existing market slice. |
| 04:40.19 | Primer | he just now told me this |
| 04:40.35 | theblue | Could someone help me think up a dialplan? |
| 04:40.51 | ManxPower | glm2k, Correct. |
| 04:41.04 | ManxPower | I just do Voip implimentaitons for companies, I donl sell mins |
| 04:41.28 | ManxPower | Ugh. Must be getting late, my typing is sucking more than usual. |
| 04:41.36 | glm2k | hehe |
| 04:41.40 | iCEBrkr | Hrrm. Can Asterisk detect the 3 tones that indicated a disconnected number? |
| 04:41.45 | *** join/#asterisk wasim_ (n=wasim@pdpc/supporter/active/wasim) |
| 04:42.07 | Jenna | anyway market or no market. Wat constitutes of complete (on a small scale) VOIP insfrastructure apart from an Asterisk gateway thingy ? |
| 04:42.07 | dca | Manx, who do you use for inboud DID's? |
| 04:42.20 | dca | or better, who do you suggest your clients use? |
| 04:42.38 | iCEBrkr | that reminds me, I gotta try NuFone |
| 04:43.41 | ManxPower | iCEBrkr, not very well. |
| 04:44.01 | ManxPower | iCEBrkr, but it's only an issue if you have an analog interface to the PSTN. |
| 04:44.14 | ManxPower | dca, the telco |
| 04:44.21 | dca | lol |
| 04:44.29 | ManxPower | dca, For my personal stuff Teliax and Nufone. |
| 04:44.41 | ManxPower | But for my clients, we don't need numbers in far away places and use local telco connections |
| 04:44.51 | iCEBrkr | Shucks |
| 04:44.54 | dca | price aside, makes sense |
| 04:45.14 | iCEBrkr | Is there any sort of signaling on a PRI for those tones? |
| 04:45.36 | ManxPower | iCEBrkr, no, you get a disconnect cause code to indicate the status of the call. |
| 04:45.42 | ManxPower | I think 37 is "disconnected" |
| 04:45.51 | ManxPower | google for "pri cause code" |
| 04:45.55 | iCEBrkr | Cool |
| 04:45.59 | iCEBrkr | That'll work for me. |
| 04:46.03 | iCEBrkr | I mean, if it works :P |
| 04:46.08 | ManxPower | iCEBrkr, it does. |
| 04:46.17 | iCEBrkr | Cool |
| 04:49.08 | Jenna | <PROTECTED> |
| 04:50.35 | Inv_arp | Jenna: not sure what yer askin |
| 04:51.20 | *** part/#asterisk AaronP (n=Aaron@c-24-6-133-255.hsd1.ca.comcast.net) |
| 04:51.20 | Jenna | Inv_arp: just want to setp an VOIP business around asterisk. what stuff would I be requiring . the infrastructure ie. |
| 04:52.38 | Inv_arp | Jenna: what type of business, you wanna be a proviser or somethin? |
| 04:52.47 | Inv_arp | provider* |
| 04:53.04 | Jenna | Inv_arp: yup . provider |
| 04:53.11 | Ikarus | Jenna: usually SER + Asterisk |
| 04:53.17 | Ikarus | but Asterisk alone also works |
| 04:53.41 | Inv_arp | Jenna: yea my provider uses SER... |
| 04:54.07 | Jenna | Ikarus: SER ? whats that ? |
| 04:54.14 | Ikarus | Jenna: SIP Express Router |
| 04:54.15 | Jenna | Inv_arp: SER ? whats that ? |
| 04:54.25 | Ikarus | Jenna: a SIP only proxy/routing thing |
| 04:54.34 | Jenna | how must does it cost |
| 04:54.38 | Ikarus | Jenna: 0 |
| 04:54.49 | Inv_arp | Jenna: SP express router |
| 04:54.51 | Ikarus | Jenna: it is mainly used to setup internal calls and load balance over multiple Asterisk media/voicemail servers |
| 04:54.55 | Inv_arp | SIP* |
| 04:55.11 | Jenna | sweet. |
| 04:55.35 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 04:55.38 | Jenna | could u guys point me to some doc regarding all this setup thing. |
| 04:55.44 | Ikarus | also SER scales better in subscribe messages for SIP |
| 04:55.57 | Primer | so can someone please tell me what's the difference between asterisk's skinny vs. sccp support? |
| 04:55.59 | patpatnz | Is there any way to access the Remote Peer ID of a sip channel from the dialplan? |
| 04:56.01 | Primer | is sccp not skinny? |
| 04:56.06 | Ikarus | Jenna: the voip wiki for Asterisk also has SER info on it |
| 04:56.17 | Jenna | I already have cell phone business running. Im hoping to extend make use of VOIP thingy |
| 04:56.20 | Ikarus | Jenna: as said, SER + Asterisk is a really common setup |
| 04:56.42 | Inv_arp | Jenna: voip-info.org |
| 04:56.45 | patpatnz | Jenna, I'm setting one up now (SER + Asterisk) |
| 04:57.07 | patpatnz | Who knows about RPIDs? |
| 04:57.55 | Jenna | Ikarus: Inv_arp: patpatnz: okay thanx guys |
| 04:58.00 | Ikarus | Right, I am going to go to bed again for a bit stomach ache making me feel like hell |
| 05:01.47 | patpatnz | I want to use the RPID instead of the SIP from address when making H323 calls |
| 05:01.52 | brc_ | hi |
| 05:02.03 | patpatnz | Does anyone know how to do that? |
| 05:02.16 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 05:04.57 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com) |
| 05:10.59 | FuriousGeorge | my voicemail seems kinda quite. |
| 05:14.39 | FuriousGeorge | and meetme doesnt meet anyone |: |
| 05:16.32 | *** join/#asterisk [dwC] (n=dwc@S0106002078e03b3f.vs.shawcable.net) |
| 05:16.39 | [dwC] | anyone use DIDx here? |
| 05:18.05 | SkramX | I dont. |
| 05:18.08 | SkramX | :( |
| 05:18.45 | FuriousGeorge | me niether |
| 05:19.01 | [dwC] | I am just trying to setup their free DID's for the first time with no luck |
| 05:19.11 | FuriousGeorge | this damn thing tells me the pin number is invalid when i clearly didnt specify a pin in meetme.conf |
| 05:19.12 | [dwC] | getting IAX rejections on my * |
| 05:19.13 | FuriousGeorge | what gives |
| 05:19.44 | *** part/#asterisk neil_ablang (n=neil@202.124.128.39) |
| 05:20.47 | theblue | Hi all. |
| 05:20.54 | theblue | Can anyone help me think of a sane dialplan? |
| 05:21.08 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 05:21.10 | FuriousGeorge | exten=>._,1,hangup |
| 05:21.14 | theblue | ... |
| 05:21.15 | patpatnz | What was that site where you could register and call for free as long as you also provided a dialout to some numbers? |
| 05:21.20 | theblue | www.fwdout.com |
| 05:21.20 | blitzrage | for those of you who have prioritized traffic using a Linux "router", what applications / software did you use? I'm looking for something that isn't going to require me to spend several weeks to set it up (I have a weekend) - I want to prioritize voice traffic over everything, and then prioritize certain subnets over others for data |
| 05:21.32 | patpatnz | theblue, ta :) |
| 05:21.38 | FuriousGeorge | blitzrage: ipcop. do it |
| 05:21.50 | blitzrage | FuriousGeorge: thanks -- will look into that |
| 05:21.52 | theblue | patpatnz: np. |
| 05:22.03 | FuriousGeorge | theblue: what do you want your dialplan to do |
| 05:22.13 | patpatnz | theblue, don't suppose you know anything about RPIDs? |
| 05:22.19 | patpatnz | :) |
| 05:22.19 | theblue | patpatnz: Sorry, no. |
| 05:22.31 | patpatnz | theblue, ah, well, was worth asking |
| 05:22.31 | theblue | FuriousGeorge: May I explain in an /msg? |
| 05:22.40 | patpatnz | anyone else know about RPIDs? |
| 05:22.43 | FuriousGeorge | theblue: sure |
| 05:22.46 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-157-68.dsl.irvnca.pacbell.net) |
| 05:23.36 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:24.11 | brc_ | hi |
| 05:24.39 | patpatnz | brc_, hi |
| 05:27.07 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 05:29.43 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 05:30.20 | *** join/#asterisk dos000 (n=dos000@i216-58-27-16.cybersurf.com) |
| 05:35.05 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 05:38.48 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 05:38.48 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 05:40.58 | *** join/#asterisk websae (i=websae@207-118-145-120.dyn.centurytel.net) |
| 05:41.18 | websae | I was curious if anyonne has an idea why i keep getting this message Oct 5 22:36:37 NOTICE[12756]: chan_sip.c:10614 handle_request_register: R |
| 05:41.18 | websae | egistration from '"BRANDO" <sip:203@206.132.218.42>' failed for '207.118.145.120 |
| 05:41.18 | websae | ' - Not a local SIP domain |
| 05:42.06 | websae | why would it say not a local SIP domain |
| 05:42.49 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
| 05:42.50 | websae | /msg nick 125 do you know what this means Oct 5 22:36:37 NOTICE[12756]: chan_sip.c:10614 handle_request_register: R |
| 05:42.50 | websae | egistration from '"BRANDO" <sip:203@206.132.218.42>' failed for '207.118.145.120 |
| 05:42.50 | websae | ' - Not a local SIP domain |
| 05:42.55 | blitzrage | websae: what version are you running? |
| 05:43.04 | websae | cvs head |
| 05:43.25 | blitzrage | websae: domains just got implemented -- it means that the far end is trying to register to a domain which you don't have configured, thus its being rejected |
| 05:43.34 | blitzrage | websae: its a "security" feature |
| 05:44.02 | websae | no...i was using the server's ip address |
| 05:44.17 | websae | the server's static ip address...as the inbound/outbound proxy |
| 05:47.15 | Primer | man, I almost have this 7920 working with chan_sccp2 |
| 05:51.59 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 05:51.59 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 05:57.08 | Primer | damn, I'm -> <- this close to getting chan_sccp2 working...but the sccp client is behind NAT and asterisk is sending the RTP to the RFC1918 address |
| 05:57.32 | snitt | loll |
| 05:58.33 | PoWeRKiLL | How not to pass audio when dialing SIP -> * -> ZAP ? |
| 05:59.50 | *** join/#asterisk asterisk99 (n=chatzill@modemcable169.194-130-66.mc.videotron.ca) |
| 05:59.58 | blitzrage | Primer: guess there's no nat=yes for sccp2 eh? |
| 06:00.15 | Primer | nope |
| 06:00.22 | blitzrage | Primer: so that it knows to pass it to the address in the IP header instead... |
| 06:00.44 | asterisk99 | Is anyone here using a TE110P (single-span T1/E1) card or the 2-3-4 span variants? i have PRI configuration problems |
| 06:01.57 | *** join/#asterisk websae (i=websae@207-118-145-120.dyn.centurytel.net) |
| 06:04.00 | ManxPower | Primer, You REALLY like pain don't you? |
| 06:04.05 | FuriousGeorge | i cannot freakin believe theyre gonna start vinny this weekend. give bolinger a chance |
| 06:04.07 | blitzrage | PAIN! |
| 06:04.12 | FuriousGeorge | not like he chucked a pick |
| 06:04.21 | blitzrage | FuriousGeorge: cricket? |
| 06:04.26 | Primer | ManxPower: it seems that if chan_sccp2 supported NAT, this would work |
| 06:04.28 | ManxPower | Is FuriousGeorge speaking english? |
| 06:04.34 | Primer | but it seems that sccp isn't nat friendly either |
| 06:04.38 | ManxPower | Primer, I doubt it supports NAT. |
| 06:04.47 | FuriousGeorge | blitzrage: sorry i thought i was in #nyjets |
| 06:04.48 | ManxPower | After all SCCP doesn't support any authentication |
| 06:04.56 | blitzrage | FuriousGeorge: :) |
| 06:06.06 | FuriousGeorge | blitzrage: continuing our conversation, at the heart of ipcop is iptables i believe so if it does QoS and doesnt support it on a per zone basis through the httpd gui, i assume you could do it by hand |
| 06:06.09 | FuriousGeorge | but what do i know |
| 06:06.12 | Primer | ManxPower: well, I imagine if the engineer that implemented SCCP on the 7920 can't figure it out, I won't be able to either |
| 06:06.37 | blitzrage | FuriousGeorge: more than likely -- good thing I'm bringing my Linux Firewalls book :) |
| 06:06.55 | Dr_Ray | I'm going to make my own asterisk fork |
| 06:07.06 | FuriousGeorge | im sure they'll be someone there to help out, works case scenario |
| 06:07.06 | Dr_Ray | it seems all the rage |
| 06:07.08 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:07.18 | FuriousGeorge | Dr_Ray: call it # |
| 06:07.26 | blitzrage | Dr_Ray: lol |
| 06:07.37 | blitzrage | ManxPower: too bad you can't make it... |
| 06:07.50 | FuriousGeorge | what is that symbol called again? is it always hash or pound? |
| 06:08.26 | FuriousGeorge | or is there another name |
| 06:08.37 | ManxPower | blitzrage, My internet connection is via WiFi to an antenna on top of a water tower. |
| 06:09.01 | FuriousGeorge | ManxPower: when you flush does it get hot? |
| 06:09.07 | ManxPower | FuriousGeorge, Octothorp (correct), Pound (USA), Hash (British) |
| 06:09.24 | Dr_Ray | tic-tac-toe thingy (my mother) |
| 06:09.31 | ManxPower | At least I didn't have to launch weather baloons to get internet access |
| 06:09.35 | Octothorp | this moment has changed me forever |
| 06:10.17 | ManxPower | I'm now calling the town I live in "Limping Muel TX", since it's not good enough even to be a "one horse town" |
| 06:10.46 | ManxPower | Mule, even |
| 06:10.54 | blitzrage | ManxPower: Octothorp is a made up term! :) |
| 06:11.15 | *** join/#asterisk criptos (n=criptos@201.145.229.189) |
| 06:11.17 | blitzrage | ManxPower: which we talk about in "thebook" :) |
| 06:11.29 | criptos | :) |
| 06:11.31 | ManxPower | blitzrage, nifty. |
| 06:11.34 | ManxPower | Like "splat"? |
| 06:11.41 | blitzrage | ManxPower: sure :) |
| 06:11.58 | ManxPower | rm splat dot splat |
| 06:12.02 | blitzrage | why is it I always feel its necessary to bring a ton of clothes with me when I travel... |
| 06:12.04 | FuriousGeorge | Number sign |
| 06:12.04 | FuriousGeorge | From Wikipedia, the free encyclopedia. |
| 06:12.04 | FuriousGeorge | (Redirected from Octothorp) |
| 06:12.11 | *** part/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
| 06:12.24 | ManxPower | blitzrage, go to more clothing optional places. |
| 06:12.33 | blitzrage | FuriousGeorge: I do like the term octothorpe though |
| 06:12.38 | ManxPower | It really saves on the laundry |
| 06:12.39 | blitzrage | ManxPower: uhhh... nah :) |
| 06:12.47 | patpatnz | anyone know about RPIDs and how/if you can access them from the dialplan? |
| 06:13.02 | FuriousGeorge | whats an rpid? |
| 06:13.09 | patpatnz | remote party id |
| 06:13.13 | *** join/#asterisk gres (n=serg@212.113.111.65) |
| 06:13.16 | patpatnz | used in sip |
| 06:13.23 | patpatnz | it's like callerid for sip I guess |
| 06:14.05 | patpatnz | anyway, doesn't look like it so I'm off, laters |
| 06:14.22 | FuriousGeorge | patpatnz: isnt it a global var ${CALLERID} |
| 06:14.30 | FuriousGeorge | lol |
| 06:15.05 | Dr_Ray | I wish I could get off work |
| 06:15.09 | Dr_Ray | to go |
| 06:15.27 | FuriousGeorge | glad ur not my dr |
| 06:15.45 | blitzrage | pattieja: FYI -- http://bugs.digium.com/view.php?id=2471 |
| 06:15.47 | Dr_Ray | I'm not actuaklly a doctor |
| 06:15.59 | blitzrage | just plays one on IRC |
| 06:16.05 | blitzrage | (I know... lame joke) |
| 06:16.06 | Dr_Ray | not so much anymore |
| 06:16.08 | Dr_Ray | er |
| 06:17.03 | Dr_Ray | Work declined to pay my way to astericon this year, why? our PBX works :) |
| 06:17.16 | blitzrage | lol |
| 06:17.20 | blitzrage | Dr_Ray: should have broke it :) |
| 06:18.52 | Dr_Ray | they bought me TDM cards and channel banks, so I should be nice |
| 06:19.22 | *** part/#asterisk Uberbot (n=Uberbot@69.252.219.76) |
| 06:19.33 | FuriousGeorge | nite all |
| 06:19.40 | Dr_Ray | so do we know who the braintrust behind OpenPBX.org is? |
| 06:19.55 | blitzrage | Dr_Ray: anthm and bkw_ |
| 06:19.58 | blitzrage | mostly |
| 06:20.06 | Dr_Ray | oh |
| 06:20.33 | Dr_Ray | I'll be quiet then |
| 06:20.38 | blitzrage | :) |
| 06:21.05 | blitzrage | meh... I don't *really* get it, but I'm sure someone will try to explain it to me; but I don't really care enough |
| 06:21.17 | Dr_Ray | if I ahd to guess it's the GPL thing |
| 06:21.28 | blitzrage | its politics -- thats about it |
| 06:23.05 | oej | Blitzrage: Morning |
| 06:25.16 | ManxPower | oej, I have to replace my cordless phones. Every time I use one of them my jitter goes off the scale |
| 06:25.48 | blitzrage | oej: !!!!!!!!!!!!!!!!!!!!! |
| 06:25.53 | blitzrage | oej: good morning! |
| 06:25.59 | ManxPower | Telecom should be run over wires, the way god intended it! |
| 06:26.14 | oej | blitzrage: Ready for a long flight? |
| 06:26.21 | blitzrage | oej: no.... :) |
| 06:26.33 | blitzrage | oej: well... I'm packed... just not looking forward to the border crossing |
| 06:26.49 | oej | ManxPower: Yes, wireless is just a scam. Throw your cell at the water tower and go get a real copper wire. Or a barbwire and run vdsl on it. |
| 06:27.20 | ManxPower | oej, I'm 20,000 ft from the CO |
| 06:27.23 | Dr_Ray | our zyxel rep is trying to get us to pitch adsl for vdsl |
| 06:29.13 | websae | anyone know why when i call someone...they can hear me....i can't hear them....i have nat=1 (my ip phone is behind a firewall) |
| 06:29.28 | websae | i mean behind a router |
| 06:29.56 | Dr_Ray | over sip? |
| 06:30.58 | ManxPower | websae, is the Asterisk server behind NAT? |
| 06:31.34 | websae | yes |
| 06:31.39 | websae | no |
| 06:31.43 | websae | asterisk is on public static ip |
| 06:31.49 | websae | my phone is behind a router |
| 06:32.25 | ManxPower | Unless the NAT router is doing packet filtering, you should only need nat=yes and qualify=yes in the sip.conf section for that device. |
| 06:33.09 | websae | qualify does what? |
| 06:33.20 | blitzrage | checks the latency / status of a device |
| 06:33.22 | ManxPower | keeps enough traffic happening so the NAT router doesn't close the connection |
| 06:33.27 | blitzrage | that too |
| 06:33.32 | blitzrage | keeps a connection open basically |
| 06:33.47 | ManxPower | Oct 6 01:33:04 NOTICE[977]: chan_iax2.c:7023 socket_read: Peer 'NuFone' is now TOO LAGGED (3377 ms)! |
| 06:33.47 | websae | hrm |
| 06:33.50 | ManxPower | damn wireless |
| 06:33.54 | mthem | voip over wireless is not a prob, not talking bout cell here |
| 06:34.20 | mthem | what system are you running? |
| 06:34.28 | mthem | wireless that is |
| 06:35.18 | mthem | we have 15 client asterisk servers backhauling to our core, sound is great... modem/fax not ok |
| 06:35.26 | websae | when i make a call.....i can hear the ring.....person picks up phone---they can hear me, i can't hear them..... |
| 06:35.38 | mthem | u running sip? |
| 06:35.41 | websae | yes |
| 06:35.42 | websae | sip |
| 06:35.50 | ManxPower | mthem, whatever the community wireless network is using |
| 06:35.53 | mthem | set the adaptor to DMZ on your soho |
| 06:36.17 | websae | nat worked before with a test server i had at another location while i was behind a router |
| 06:36.31 | mthem | ManxPower: so a standard, 802.11 something, ya there is not much QoS there |
| 06:36.48 | ManxPower | mtgh, *nod*, but it's the only internet access I have. |
| 06:37.01 | mthem | u running IAX? |
| 06:37.06 | mthem | over the wireless |
| 06:37.09 | ManxPower | mtgh, of course. |
| 06:37.34 | mthem | .. of course.. well, dont know mush about that, |
| 06:37.55 | websae | anyone have any ideas...i have nat=yes and qualify=yes.....when i make call...person can hear me, i can't hear them though... |
| 06:38.02 | ManxPower | If all else fails I'll just get a couple of more phone lines |
| 06:38.06 | websae | i can hear the phone ringing too |
| 06:38.10 | websae | any help would be appreciated |
| 06:38.11 | ManxPower | websae, your router is filtering packets. |
| 06:38.25 | mthem | websae: just try the DMZ to make sure the FW is not the prob |
| 06:38.31 | websae | i did |
| 06:38.32 | websae | DMZ |
| 06:38.33 | ManxPower | websae, your device generates the ringing, you are not hearing the far end's ringing |
| 06:38.41 | mthem | websae: make sure SPI is not on |
| 06:38.49 | websae | SPI? |
| 06:38.52 | mthem | websae: if your router supports ti |
| 06:39.04 | mthem | Stateful Packet Inspection |
| 06:39.09 | websae | things worked fine before behind to router using a test server at a remote location |
| 06:39.20 | websae | but with this other server...not working to make calls where they can hear me |
| 06:39.43 | mthem | ii hear u, but its not working now, so you should start with your FW |
| 06:40.01 | mthem | you on Nufone? |
| 06:40.22 | mthem | whats the server IP? |
| 06:40.23 | websae | ? |
| 06:40.35 | mthem | who do u terminate with? |
| 06:40.44 | websae | 206.132.218.42 |
| 06:40.46 | websae | private carrier |
| 06:40.58 | mthem | sure but who? |
| 06:41.07 | websae | comsolo |
| 06:41.26 | websae | the server keeps registering my phone's private ip address |
| 06:41.31 | websae | how do i get it so it doesn't do that? |
| 06:41.36 | websae | that could be messing things up |
| 06:41.48 | ManxPower | websae, then the nat=yes is not being read |
| 06:42.06 | ManxPower | is "sip show peers" show the private IP instead of the publicv IP then that's what's happening. |
| 06:42.17 | ManxPower | Either than or your SIP client or your router is doing something nasty |
| 06:42.23 | websae | what can i do about that |
| 06:42.28 | ManxPower | websae, fix it |
| 06:42.36 | ManxPower | websae, what router are you using? |
| 06:42.45 | websae | a cheap netgear |
| 06:42.45 | ManxPower | What SIP device are you using? |
| 06:42.52 | websae | sipura 841 hard phone |
| 06:42.57 | mthem | could olso be that the new server does not support contin. sessions in its FW |
| 06:43.10 | ManxPower | websae, and you have all the NAT options in the Sipura turned off? |
| 06:43.25 | lehel | people after a few secs my iax connection is muted.. i ran an iax2 debug: >> Tx-Frame Retry[-01] ...>> Tx-Frame Retry[000] .. >> Rx-Frame Retry[ No], i can't find where is the bug |
| 06:43.42 | websae | should i? |
| 06:43.44 | websae | have them turned off? |
| 06:43.46 | ManxPower | websae, Yes. |
| 06:43.47 | websae | in my sipura? |
| 06:43.53 | ManxPower | Of course. |
| 06:44.09 | websae | nat mapping = no? |
| 06:44.14 | websae | that's what i have |
| 06:44.44 | ManxPower | Except for the NAT Keep Alive Enable: option. If you set that option to yes, then you don't need qualify=yes |
| 06:45.03 | ManxPower | anyway, off to bed |
| 06:46.05 | websae | hrm |
| 06:46.47 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:47.09 | mthem | websae: do u have access to the reg. server? |
| 06:48.03 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:48.04 | websae | yeah |
| 06:48.06 | websae | yes i do |
| 06:48.08 | websae | have access |
| 06:48.25 | mthem | what does your iptables look like (if thats what you are running |
| 06:49.00 | websae | what does iptables have to do with it? |
| 06:49.12 | websae | i can make the call....establish a call...they can hear me..i can't hear them |
| 06:49.13 | blitzrage | night all -- off to Anaheim tomorrow! |
| 06:50.03 | websae | mthem: cisco phone works fine |
| 06:50.16 | mthem | right that indicates that 5060 is open, since the call is setup.... but after taht the server picks a random udp port for the voice |
| 06:50.31 | mthem | from behind the same FW? |
| 06:51.04 | websae | so what should i do? |
| 06:51.19 | mthem | is the cisco working from behind the same router? |
| 06:51.37 | websae | behind a different router |
| 06:52.07 | websae | set to DMZ |
| 06:52.12 | websae | works fine |
| 06:52.37 | mthem | ok, try change your ip on the sipura.... seen that work before |
| 06:52.47 | mthem | dont know why exactly |
| 06:53.15 | websae | change my ip? |
| 06:53.17 | websae | how so? |
| 06:53.22 | websae | on my sipura? |
| 06:53.42 | mthem | right |
| 06:54.01 | mthem | just another ip from your LAN pool |
| 06:54.37 | mthem | just to make sure is the cisco reg. to the same server you are trying now? |
| 06:57.48 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 06:58.53 | mthem | websae: u live? |
| 06:58.58 | websae | yeah |
| 06:59.00 | websae | i am here |
| 06:59.04 | websae | maybe set it static? |
| 06:59.34 | mthem | u have to otherwise you will get the same from the router DHCP |
| 07:00.34 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:01.11 | *** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 07:01.45 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:01.52 | websae | ok |
| 07:01.53 | websae | no go |
| 07:02.54 | websae | same issues |
| 07:03.08 | FuriousGeorge | websae: put everything in the dmz |
| 07:03.23 | websae | yeah i have my router dmz set on |
| 07:03.26 | websae | for the sipura phone |
| 07:03.28 | FuriousGeorge | is something blocking rtp 10000-20000 |
| 07:03.35 | websae | nope |
| 07:03.43 | FuriousGeorge | i got nuthin |
| 07:04.14 | FuriousGeorge | want me to try and log my client into ur server (not in a gay way) |
| 07:04.44 | FuriousGeorge | ok, not funny |
| 07:04.46 | FuriousGeorge | :) |
| 07:04.51 | FuriousGeorge | :| |
| 07:05.02 | FuriousGeorge | tough crowd |
| 07:05.24 | mthem | ya that is a 011871-5 call worth ;) |
| 07:05.59 | mthem | INMARSAT - Atlantic East |
| 07:06.05 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 07:06.05 | mthem | $12 per min |
| 07:06.30 | *** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net) |
| 07:06.33 | FuriousGeorge | wow, any names of any places i might know |
| 07:06.41 | *** join/#asterisk protien (i=jjmtrev@203-173-26-187.dyn.iinet.net.au) |
| 07:06.44 | mthem | >>? |
| 07:06.57 | mthem | ur the joker, better comebacks |
| 07:07.02 | mthem | ;) |
| 07:07.14 | FuriousGeorge | take my wife please |
| 07:07.24 | FuriousGeorge | seriously though |
| 07:08.09 | mthem | ya? |
| 07:08.12 | FuriousGeorge | where does INMARSAT cover? |
| 07:08.29 | mthem | east atlantic.... or west atlantic |
| 07:08.37 | mthem | ..... hehe its right there |
| 07:09.04 | mthem | maybe this is better, water |
| 07:09.15 | FuriousGeorge | <PROTECTED> |
| 07:09.21 | mthem | ;) thanks |
| 07:10.00 | FuriousGeorge | is that what goes on here at 3EST |
| 07:10.31 | FuriousGeorge | thats better |
| 07:11.07 | mthem | websae: u never answered if the cisco was reg. to the same server u where trying? |
| 07:13.50 | mthem | dont anyone have an interesting problem tonight? |
| 07:13.54 | FuriousGeorge | i got the wierdest issue. something is trumping the values i set in meetme.conf, and asking me for a pw when there is none |
| 07:14.11 | FuriousGeorge | or telling me its wrong when its not, man |
| 07:15.08 | *** join/#asterisk mbranca (n=matteo@host-210-mi.espia-net.net) |
| 07:15.34 | mthem | hows your DTMF working in other calls? |
| 07:15.47 | mthem | or how are you sending your DTMF |
| 07:16.01 | FuriousGeorge | and the wierdist part is that voip info says asterisk doesnt even need to be restarted for meetme.conf changes |
| 07:16.04 | FuriousGeorge | inband, ulaw |
| 07:16.21 | FuriousGeorge | actually my remote client is gsm! thats where im testing from |
| 07:16.24 | FuriousGeorge | but |
| 07:16.43 | mthem | so in-audio over GSM? |
| 07:16.50 | FuriousGeorge | why is it asking me for a pw when i specify nothing but the exten |
| 07:17.28 | FuriousGeorge | actually, now that i look at sip.conf, im not passing it inband |
| 07:17.28 | mthem | ..ya... dunno much about conf. |
| 07:18.33 | FuriousGeorge | is there anyway to use the time to randomize in the dialplan |
| 07:19.19 | mthem | randomize? |
| 07:19.47 | FuriousGeorge | mthem: i was totally thinking about make festival say random wierd shit, man |
| 07:21.15 | FuriousGeorge | but the math operators are only +-/* |
| 07:21.19 | mthem | hows that working for u? |
| 07:21.34 | FuriousGeorge | no sqrt or ^ |
| 07:21.46 | mthem | the festival text/voice |
| 07:22.04 | FuriousGeorge | no, the random part |
| 07:22.15 | FuriousGeorge | oh festival, for me |
| 07:22.19 | mthem | i wanted to get my emails as voice... but dident want to spend the time if it sucked |
| 07:22.38 | mthem | ya |
| 07:22.40 | FuriousGeorge | mthem: i know the mail.conf can disregard short messages |
| 07:22.47 | FuriousGeorge | or voicemail.conf i mean |
| 07:23.12 | mthem | FuriousGeorge: now you are saying random stuff |
| 07:23.29 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 07:23.35 | FuriousGeorge | i guess i am |
| 07:23.42 | FuriousGeorge | i see what you mean |
| 07:23.47 | Zeeek | Morgen |
| 07:24.20 | mthem | morgen, er det ikke lidt tidligt du er oppe? |
| 07:24.34 | FuriousGeorge | the quality is pretty terrible, to answer your question |
| 07:24.39 | Zeeek | ummmmmmmm.... ya? |
| 07:25.23 | mthem | oh, Zeeek, morgen is Good Morning in Danish |
| 07:25.29 | mthem | ;) |
| 07:25.42 | Zeeek | yes but I didn't jknow what fololwed |
| 07:25.44 | mthem | <FuriousGeorge>,thought so |
| 07:25.57 | Zeeek | speaking of Denmark... |
| 07:26.38 | FuriousGeorge | do danes understand the dutch better worse or equally as well as the dutch comprehend the duetch |
| 07:26.53 | FuriousGeorge | how do you spell doy-iich |
| 07:27.10 | FuriousGeorge | ? |
| 07:27.39 | mthem | <FuriousGeorge>: dutch is alot more lkike german, and completely not understandeble ulaw or not |
| 07:27.59 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 07:28.02 | FuriousGeorge | and dane is more scandenavian |
| 07:28.04 | FuriousGeorge | ? |
| 07:28.17 | FuriousGeorge | danish |
| 07:28.19 | FuriousGeorge | imean' |
| 07:28.28 | mthem | <FuriousGeorge>: norway sweden denmark, ya we get eachother |
| 07:28.32 | *** join/#asterisk uter (n=fn@213.178.78.120) |
| 07:28.40 | FuriousGeorge | i get the portuguese |
| 07:28.53 | FuriousGeorge | but i get the gallegos better |
| 07:29.04 | Zeeek | dansk |
| 07:29.10 | mthem | hehe, right |
| 07:29.20 | uter | moin |
| 07:29.22 | FuriousGeorge | ur welcome? |
| 07:29.26 | FuriousGeorge | good? |
| 07:29.31 | Zeeek | hey you danskers, are your browsers set to dansk language? |
| 07:29.32 | lehel | me: nl, hu, es ;)P |
| 07:29.51 | mthem | the guys r speaking dialect danish |
| 07:29.51 | Zeeek | if so, look here and tell me if it is in Danish : http://blog.chateau-palmer.com/ |
| 07:30.12 | FuriousGeorge | csa lkj eqr |
| 07:30.29 | Zeeek | hüsker dü ! |
| 07:30.29 | FuriousGeorge | later all |
| 07:30.47 | Zeeek | except I think that's norwegian |
| 07:31.00 | mthem | Zeeek: its in english... go to bed |
| 07:31.03 | mthem | ;) |
| 07:31.12 | Zeeek | no look at the interface, not the articles |
| 07:31.18 | Zeeek | there are two articles in dansk |
| 07:31.27 | gordonjcp | there aren't any proper Scots Gaelic translations |
| 07:32.42 | mthem | Zeeek: ya thats danish |
| 07:32.50 | Zeeek | the calendar and all, right? |
| 07:33.03 | Zeeek | the software is supposed to detect your preference |
| 07:33.39 | mthem | Zeeek: the arcives are in danish months |
| 07:33.42 | mthem | im in LA |
| 07:33.46 | Zeeek | ok gut |
| 07:34.22 | Zeeek | none of our codecs match 0x255 |
| 07:35.01 | mthem | http://babelfish.altavista.com/ |
| 07:35.06 | mthem | there u are |
| 07:35.22 | mthem | or atleast dutch |
| 07:35.40 | mthem | did all my german essays |
| 07:35.45 | *** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it) |
| 07:36.15 | mthem | hehe, dirt |
| 07:36.43 | mthem | Translate: I love to masterbate |
| 07:36.50 | mthem | nice going |
| 07:37.08 | syle | whats proper slang word for weed there |
| 07:37.15 | denon | /part * |
| 07:37.18 | Zeeek | this screen is in a public place huys. My coworkers will think I'm on a wanker channel |
| 07:37.28 | Zeeek | hey denon |
| 07:37.34 | denon | hey Zeeek |
| 07:37.53 | mthem | mmm, hold on |
| 07:38.12 | mthem | same as here |
| 07:38.14 | mthem | cronic |
| 07:38.28 | mthem | any asterisk questions? |
| 07:38.38 | Zeeek | on what basis? |
| 07:38.48 | FuriousGeorge | does asterisk support intercom/conference in any channel? |
| 07:38.57 | FuriousGeorge | sip clients implement it |
| 07:39.19 | syle | just wire your sound card |
| 07:39.21 | FuriousGeorge | but you cant force that on the user in certain instances |
| 07:39.23 | Zeeek | there is meetme |
| 07:39.29 | mthem | FuriousGeorge: I should think so but Zap is by far the easiest |
| 07:40.02 | *** join/#asterisk fenlander (n=neils@82.152.81.57) |
| 07:40.04 | FuriousGeorge | mthem: you cant force the user to answer call waiting |
| 07:40.29 | syle | hmmm |
| 07:40.31 | mthem | FuriousGeorge: no that would make no sence |
| 07:40.47 | syle | that would be a good peice of code to write that can |
| 07:40.48 | FuriousGeorge | mthem: "There's a fire" |
| 07:40.54 | syle | barge in on the line |
| 07:41.02 | FuriousGeorge | syle: i bet you can jerry rig it in the dialplan somehow |
| 07:41.11 | syle | nope |
| 07:41.16 | FuriousGeorge | isnt there a chaninterupt dialplan cmd? |
| 07:41.24 | syle | nope |
| 07:41.29 | mthem | oh, well, script that kills all channels then calls all extensions with a message |
| 07:41.43 | FuriousGeorge | mthem: still cant force them to answer |
| 07:42.04 | FuriousGeorge | i had a job once that anouncements came in over our phones (if we werent on'em) |
| 07:42.15 | mthem | even if they saw the smoke u could not force them, but that is what a free contry is all about ;) |
| 07:42.18 | syle | ok this is pissing me off where is the search wiki box on voip-info.org gone |
| 07:42.36 | FuriousGeorge | syle: pissing me of too |
| 07:42.40 | syle | this been going on since the weekend |
| 07:42.45 | FuriousGeorge | ok im really going this time |
| 07:42.57 | Zeeek | the search sucked anyway - use google |
| 07:43.06 | FuriousGeorge | syle: i thinnk it was actually down for a bit before that, but im not really in the loop |
| 07:43.34 | FuriousGeorge | the whole site, i mean |
| 07:43.37 | FuriousGeorge | gotta go |
| 07:43.50 | mthem | anyone know what the main problem is, they seem to have alot of downtime at peek hours |
| 07:44.01 | mthem | traffic? |
| 07:46.51 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 07:47.25 | uter | anyone using snom phones enjoying blinkenlights? |
| 07:48.17 | mthem | well guys im off, later |
| 07:48.37 | *** join/#asterisk Thoran (n=Thoran@p54A5A66F.dip0.t-ipconnect.de) |
| 07:49.38 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 07:51.10 | uter | on my snom360 the LEDs work fine with hints and bristuff devstate, but they don't blink |
| 07:52.37 | uter | i tried different firmwares and asterisk versions, but i don't know how to make it work |
| 07:53.24 | JamesDotCom | isnt it a param in app_devstate? |
| 07:54.12 | uter | well, all parameters i tried either switched on or switched off the led |
| 07:56.34 | uter | the only time, i see a blinking led,is, when the telephone itself is called |
| 07:57.24 | uter | but i can see no main differences in the SIP-traffic |
| 07:58.51 | uter | so i think the problem is the firmware |
| 07:59.48 | uter | i've got the latest one, but i think snom sometimes releases buggy versions |
| 08:04.05 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255) |
| 08:04.11 | MuppetMaster | Hello |
| 08:11.15 | *** join/#asterisk casio_ (i=jjhhjkh@200-126-69-179.bk5-dsl.surnet.cl) |
| 08:16.49 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:17.53 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:19.52 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:25.25 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.168.255) |
| 08:25.50 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 08:27.36 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:28.13 | RoyK | hi. with asterisks CDR logging, is a billsec counted on the start of a second, or at the completion of a second? |
| 08:29.53 | *** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) |
| 08:29.54 | Delvar | its the same thing :) start of one end of another... |
| 08:33.25 | lehel | RoyK, which billing soft do you use? astcc.. areskicc..? |
| 08:39.33 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 08:39.47 | RoyK | lehel: neither of them |
| 08:40.21 | RoyK | lehel: all i want to know is the usual CDR logs, the billsec there, is it counted from the start of the second or the completion of the second? |
| 08:42.03 | *** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com) |
| 08:43.45 | *** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8) |
| 08:46.32 | casio_ | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 08:46.56 | casio_ | i have this problem with a x100p |
| 08:47.12 | casio_ | can anybody help me? |
| 08:47.30 | casio_ | # modprobe wcfxs |
| 08:47.34 | casio_ | :( |
| 08:47.38 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 08:49.11 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 08:52.25 | RoyK | hrmf |
| 08:52.37 | RoyK | switch monkey wants me to 'divert' a call to some number |
| 08:52.53 | RoyK | wtf is a 'divert' apart from a new call with RDNIS set in the PRI SETUP? |
| 08:54.30 | *** join/#asterisk Attila_Kovacs (n=kovacsat@dsl51B6785D.pool.t-online.hu) |
| 09:05.55 | *** join/#asterisk ful|work (n=fulgas@213.58.130.46) |
| 09:11.46 | *** join/#asterisk skiold (n=userid@84-121-64-126.onocable.ono.com) |
| 09:12.28 | Spacebar | switch monkey? hahaha |
| 09:12.55 | snitt | try to spend your time on hold not thinking about the blue eyed polar bear.. |
| 09:12.57 | snitt | wtf, lol |
| 09:22.27 | *** join/#asterisk casio_ (i=jjhhjkh@200-126-69-179.bk5-dsl.surnet.cl) |
| 09:26.18 | *** join/#asterisk Bonzai070 (n=pirch@wbs-146-162-61.telkomadsl.co.za) |
| 09:28.38 | *** join/#asterisk pr0 (n=LANmower@ndn-165-130-135.telkomadsl.co.za) |
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| 09:30.08 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 09:30.15 | patpatnz | hi everyone |
| 09:31.18 | Bonzai070 | looking for asterisk guru to assist in a large scale asterisk implementation flights and acommodation paid private me if interested |
| 09:32.24 | patpatnz | Looking for someone to help me for free ;) |
| 09:32.47 | Bonzai070 | patpatnz linux is free decent help is never free |
| 09:32.55 | patpatnz | I know |
| 09:33.03 | patpatnz | Actually, thatys not really true |
| 09:33.09 | patpatnz | I seem to help a lot of people for free |
| 09:33.18 | iDunno | patpatnz: know that feeling ;) |
| 09:33.21 | patpatnz | Guess I'm a sucker |
| 09:33.30 | Zeeek | karma |
| 09:33.50 | iDunno | patpatnz: and then when you need there assisstance, they've buggered off on holiday/have just had a death in the family/are pissed/something. |
| 09:33.59 | Bonzai070 | patpatnz i help for free were i can |
| 09:34.01 | patpatnz | iDunno: yup |
| 09:34.18 | patpatnz | Well, anyone know about RPIDs then? :) |
| 09:35.01 | patpatnz | I have a asterisk acting as a SIP <-> H.323 gateway and I want to use the RPID from the sip invite as the CLI of the H.323 request |
| 09:35.46 | patpatnz | :D |
| 09:35.58 | iDunno | whatthehellisaRPID? |
| 09:36.10 | patpatnz | Remote Peer ID |
| 09:36.18 | iDunno | ahh, course. |
| 09:36.21 | patpatnz | it's like CLI for SIP |
| 09:36.36 | iDunno | right - so, erm... |
| 09:36.51 | patpatnz | mm? |
| 09:36.52 | iDunno | how about just using SetCallerID(${SIPRPID}) |
| 09:37.04 | iDunno | before doing the h323 dial. |
| 09:37.07 | patpatnz | SIPRPID doesn't exist sadly :( |
| 09:37.36 | patpatnz | I can't find out if it sets a var at all |
| 09:37.38 | iDunno | well, no, I wasn't expecting it to. I could look in the book and see what I can find... |
| 09:37.49 | patpatnz | Book?! |
| 09:38.20 | *** join/#asterisk pppau (n=ps@200.115.231.78) |
| 09:38.38 | pppau | hi |
| 09:38.46 | patpatnz | hi! |
| 09:38.58 | patpatnz | iDunno: is there a book? |
| 09:39.06 | pppau | anyone who used chan_alsa? |
| 09:39.50 | pr0 | can AMP be installed with XAMPP? |
| 09:39.54 | iDunno | there's a book. |
| 09:40.07 | pr0 | i'm really struggling my arse off here |
| 09:40.12 | patpatnz | really? whats it called? |
| 09:40.16 | iDunno | it's only really an introduction, though... but it has got some handy things in it. |
| 09:40.21 | patpatnz | from what publisher |
| 09:40.23 | patpatnz | ? |
| 09:40.26 | iDunno | Asterisk: The Future of Telephony. |
| 09:40.31 | patpatnz | I'm sure it would come in handy |
| 09:40.31 | iDunno | O'Reilly. |
| 09:40.32 | lehel | pr0.. XAMPP? |
| 09:40.41 | patpatnz | Sweet, thanks iDunno |
| 09:40.54 | patpatnz | I'll be buying it I rekon |
| 09:41.09 | pppau | I'm getting too much delay on mic input using chan_alsa, anyone knows if it is normal? |
| 09:41.26 | patpatnz | pppau: use a user client? |
| 09:41.43 | patpatnz | does anyone use those ones, chan_alsa and chan_oss?! |
| 09:42.31 | iDunno | for testing stuff it's reasonably handy. |
| 09:42.55 | patpatnz | umm |
| 09:42.58 | patpatnz | okay |
| 09:43.03 | pppau | patpatnz: yes, I need to interface to audio |
| 09:43.48 | Bonzai070 | were in asterisk do i enable callerid from my pstn to go through to my sip phones?.. |
| 09:44.17 | pppau | i call * from a SIP ata, and sound gets out fron the speakers ok, but the other way has 1 sec delay! |
| 09:44.19 | patpatnz | Bonzai070: in the config for your channel |
| 09:44.36 | pr0 | never mind... |
| 09:44.53 | patpatnz | pppau: maybe your box is overloaded and can't compress fast enough? |
| 09:45.04 | patpatnz | unless your using g711 |
| 09:45.30 | pppau | g711 |
| 09:45.45 | pppau | and only that call |
| 09:45.58 | patpatnz | hmm |
| 09:46.07 | patpatnz | what are you using it for? |
| 09:46.20 | pppau | tested it on 3 different computers, with 1.0.x and 1.2 |
| 09:46.45 | pppau | just linksys pap2 to * |
| 09:47.24 | patpatnz | but why chan_alsa? do you have something in mind for it or just for testing? |
| 09:47.41 | patpatnz | if your just doing it for testing don't worry about the delay |
| 09:48.05 | pppau | no, i need it to interface to an intercom |
| 09:48.10 | patpatnz | ah |
| 09:48.22 | patpatnz | two way? |
| 09:48.27 | pppau | any clues? |
| 09:48.59 | patpatnz | some config error maybe? |
| 09:49.09 | pppau | just Dial(Console/dsp) |
| 09:49.20 | pppau | no, all default params |
| 09:49.32 | patpatnz | umm, well, dunno :) |
| 09:49.34 | patpatnz | sorry |
| 09:49.53 | syle | yeah you could do that |
| 09:50.00 | syle | does sound card auto pickup? |
| 09:50.17 | pppau | autoanswer? yes |
| 09:50.59 | syle | yeah dial dsp then and open up one of your sip phones and wire the sound card to it for an intercom is cheapest way |
| 09:51.00 | pppau | sound quality is perfect in both directions |
| 09:52.25 | syle | well how about a cheap home theatre receiver |
| 09:52.31 | pppau | syle: not so easy, have to do some processing when answering the call |
| 09:52.32 | syle | wire sound card to that |
| 09:52.40 | syle | run speakers everywhere |
| 09:52.54 | patpatnz | syle: it's not the issue of how to do it |
| 09:53.55 | syle | what is issue? |
| 09:54.09 | patpatnz | syle: audio delay from soundcard to sip phone |
| 09:54.32 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 09:54.39 | pppau | yes 1 sec!!! and no delay in the other sense |
| 09:55.48 | pppau | just 1 sip phone calling to *, with 1 extention Dial(Console/dsp) |
| 09:56.18 | syle | and you press 1 to hangup or something |
| 09:56.27 | syle | guess you could just hangup |
| 09:56.45 | pppau | what for? |
| 09:56.58 | patpatnz | syle: what are you talking about?! |
| 09:57.13 | syle | nm |
| 09:57.14 | pppau | the problem is the audio latency |
| 09:58.09 | syle | well what kind of cheap ass phone you got |
| 09:58.20 | pppau | does anyone have * installed to test it ? |
| 09:58.36 | patpatnz | pppau: none of my * servers have soundcards |
| 09:59.15 | pppau | patpatnz: thanks anyway |
| 09:59.35 | patpatnz | pppau: it's weird though |
| 10:00.40 | pppau | yes, someone must have noticed, but no info found googling. That why I'm thinking I must be nissing something |
| 10:00.46 | *** join/#asterisk ^X-works (n=r0x0r@host111-109.pool8257.interbusiness.it) |
| 10:01.09 | patpatnz | you checked out the wiki page on voip-info.org ? |
| 10:01.59 | pppau | yes, very poor info on chan_alsa |
| 10:02.05 | patpatnz | mm |
| 10:02.17 | patpatnz | not too surprising I guess |
| 10:02.32 | patpatnz | mostly people use it as a broadcast/paging system I think |
| 10:02.37 | patpatnz | (one way) |
| 10:02.42 | pppau | but, someone must have tested it |
| 10:02.49 | *** join/#asterisk NirS (n=NirS@62.90.49.94) |
| 10:05.01 | patpatnz | ya, I guess |
| 10:05.36 | iDunno | the asterisk server running on my home box has a soundcard ;) |
| 10:05.58 | patpatnz | pppau: your the tester ;) |
| 10:06.09 | patpatnz | now submit a bug! ;) |
| 10:06.55 | patpatnz | night :) |
| 10:06.57 | *** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 10:07.35 | pppau | yes, just wanted someone else to check it, to know if I'm not making any stupid mistake |
| 10:07.40 | RoyK | playtones(snore) |
| 10:08.04 | RoyK | sleep($[ 3600 * 8 ]) |
| 10:08.07 | RoyK | stopplaytones |
| 10:08.16 | pppau | RoyK: :) |
| 10:10.00 | pr0 | has anyone successfully used amp with xampp? |
| 10:12.58 | pr0 | guess not |
| 10:13.25 | iDunno | surely you'd use Wait($[ 3600 * 8]) |
| 10:13.40 | lehel | pr0: XAMPP is an easy to install Apache Distribution for Linux < this is what are youtalking about? |
| 10:13.43 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.93) |
| 10:13.53 | pr0 | yup |
| 10:14.17 | pr0 | if I can install amp on it, my problems are resolved. |
| 10:14.24 | lehel | :P |
| 10:14.40 | pr0 | otherwise, pats php breaks this livecd. |
| 10:15.15 | *** join/#asterisk samyl (n=samyl@194.167.18.244) |
| 10:15.41 | samyl | hello |
| 10:15.58 | samyl | i've a problem with meetme |
| 10:16.32 | samyl | i've installed asterisk 1.03 with ast_data |
| 10:16.49 | samyl | i don't have a digium card |
| 10:16.53 | PoWeRKiLL | RoyK do you have a TE device ? |
| 10:17.42 | samyl | so I've installed zaptel-1.0.9.1 with option "ztdummy" |
| 10:17.43 | RoyK | PoWeRKiLL: que? |
| 10:18.04 | RoyK | PoWeRKiLL: what TE dev? terminal equipment? |
| 10:18.48 | samyl | and when I launch asterisk |
| 10:19.12 | samyl | asterisk tell me channel.c:1965 ast_request: No channel type registered for 'zap' |
| 10:19.13 | samyl | Oct 6 13:54:50 WARNING[9395]: app_meetme.c:272 build_conf: Unable to open pseudo channel - trying device |
| 10:19.48 | RoyK | samyl: dunno, but just perhaps loading chan_zap will help....... |
| 10:21.10 | *** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129) |
| 10:21.16 | fishboy1669 | hi |
| 10:21.27 | samyl | it's a silly question but how can i load this module chan_zap? |
| 10:21.34 | pppau | hi |
| 10:21.54 | fishboy1669 | hi |
| 10:21.59 | fishboy1669 | hows things |
| 10:22.22 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 10:22.23 | fishboy1669 | hi samyl |
| 10:22.27 | pppau | fishboy1669: have you used chan_alsa or chan_oss? |
| 10:22.36 | fishboy1669 | do u mean how do u load zaptel |
| 10:22.47 | fishboy1669 | as in modprobe zaptel |
| 10:22.49 | samyl | hi fishboy |
| 10:22.53 | fishboy1669 | modprobe wcfxo |
| 10:23.04 | samyl | yes i know |
| 10:23.13 | fishboy1669 | in that case i dont know |
| 10:23.15 | samyl | i did it |
| 10:23.34 | fishboy1669 | unless u have configured your zaptel.cfg incorrectly |
| 10:25.03 | samyl | i've just uncommented 2 items "loadzone" and "defaultzone" because i don't have a digium card |
| 10:25.28 | *** join/#asterisk Tili (i=Tili@202-133-67-70-dialup.sat.net.pk) |
| 10:25.41 | samyl | i use a linux kernel 2.6 |
| 10:25.51 | samyl | i've patched udev |
| 10:25.58 | samyl | but nothing |
| 10:26.05 | pppau | Tili: I have a question for you |
| 10:26.42 | PoWeRKiLL | RoyK no a Digium Card like a TE410P or something like this ? |
| 10:26.49 | Tili | pppau: ok |
| 10:29.35 | pppau | I place a call from a SIP ata to an * box with one extension Dial(Console/dsp), I hear perfectly on the speakers what i say on the phone, but in the other sense (fom mic in to phone) I have 1 sec delay! |
| 10:30.19 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 10:30.23 | pppau | Tili: did you use chan_alsa? |
| 10:32.13 | *** join/#asterisk Reputation (i=Reputati@i5387B667.versanet.de) |
| 10:32.23 | Tili | no i never use it. |
| 10:32.41 | RoyK | PoWeRKiLL: yes. we have a few TE410Ps and a couple of A104s |
| 10:32.49 | Reputation | hi, is there someone who knows what i have to do when asterisk is behind my router |
| 10:33.25 | *** join/#asterisk aor (n=bob@181-48.244.81.adsl.skynet.be) |
| 10:33.43 | RoyK | Reputation: eat nutrious food and stay away from too much alcohol |
| 10:34.09 | Reputation | and this helps? mhhhhhhhh....i should give it a try :P |
| 10:34.26 | aor | Hi, I'm trying to set up realtime with asterisk 1.2 but I keep receiving this error message : Oct 6 12:25:04 WARNING[6661]: config.c:893 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 10:34.40 | aor | I'm using the same configuration that was working with CVS-HEAD |
| 10:35.31 | aor | Do I have to install a special module in order to have rt working ? I tought this was included in the base version of * now |
| 10:37.49 | *** join/#asterisk antoniofcano (n=antonio@5.Red-80-32-90.staticIP.rima-tde.net) |
| 10:37.53 | antoniofcano | Hi all |
| 10:38.07 | Reputation | hi |
| 10:38.10 | *** join/#asterisk christo (n=chris@195.82.114.14) |
| 10:38.15 | antoniofcano | One little question |
| 10:38.16 | christo | morning all |
| 10:40.07 | antoniofcano | I've got two SIP trunks with the same provider and the register is made from the same Asterisk machine, the problem is that only one of the sip trunks goes fine :-/. Could it be due to both uses the same port to register |
| 10:40.10 | Reputation | when i use * in my litte network with xlite everything works fine. people from internet can also connect to my *. but we can't hear us when we call. |
| 10:41.10 | *** join/#asterisk cjk (n=cjk@212.233.32.175) |
| 10:41.12 | Reputation | so i guess that i have to forward some ports or stuff like that? do you know what i have to do? |
| 10:41.33 | Delvar | Reputation: canreinvite=no ? |
| 10:41.44 | Reputation | yes |
| 10:41.54 | antoniofcano | Reputation, it is maybe a NAT. Try to NAT = yes |
| 10:41.57 | *** join/#asterisk Teeli (i=Tili@202-133-67-90-dialup.sat.net.pk) |
| 10:42.08 | *** join/#asterisk wundaboy (n=asdf@67.189.30.47) |
| 10:42.23 | Reputation | ii allready have canreinvite=no and nat=yes |
| 10:42.38 | johnm | can anyone confirm the way to set CallerID in * HEAD? |
| 10:42.38 | antoniofcano | externip=public_ip and localnetwork=... |
| 10:42.39 | antoniofcano | ¿? |
| 10:42.55 | christo | Reputation - are you behind firewalls? you need ports 5004 5060 tcp/udp afaicr |
| 10:43.45 | Reputation | i'm behind a router, but i deactivated the firewall. |
| 10:44.22 | Delvar | portforwarded 5060 and 10000-20000 udp to asterisk? |
| 10:44.35 | Delvar | set in sip.conf local noetowrk and extenal ip? |
| 10:44.43 | christo | Reputation - I also have found with x-lite that it's slow to pick up changes to it's config.. close the phone app, then right click on the icon in the system tray and select exit (that bit's important) then try re-opening x-lite. Sometimes that jolts it into action |
| 10:45.10 | Reputation | i forwardet 5060 to asterisk and 10000-20000 to my client pc. do i have to forward it to asterisk? |
| 10:45.21 | Delvar | yes |
| 10:45.41 | Delvar | 10000 to 20000 are RTP to asterisk (see the range in rtp.conf) without this you wont get audio |
| 10:45.59 | Delvar | o_0 |
| 10:46.02 | Reputation | ok. i hope that this solves my problem. thanks ;-) |
| 10:46.28 | Reputation | bye...maybe i'll see you later again :) |
| 10:46.50 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 10:47.17 | christo | I just recompiled libpri and libzaptel to newer versions, but I can't rmmod zaptel.. it says 'Device or resource busy'. Is there a way around this? |
| 10:47.28 | christo | I just want to reload the module with a new modprobe.. |
| 10:47.42 | *** part/#asterisk Reputation (i=Reputati@i5387B667.versanet.de) |
| 10:48.26 | Delvar | you have to stop asterisk |
| 10:48.46 | Delvar | and rmmod anything that uses zaptel like wcfxx... |
| 10:48.51 | Delvar | ztdummy etc... |
| 10:49.42 | christo | thanks Delvar |
| 10:49.55 | christo | I was rmmodding them the wrong way round :) |
| 10:50.07 | Delvar | hehe np |
| 10:51.55 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 10:52.00 | queuetue | Hi. Can anyone explain what an "407 Proxy Authentication Required" error means? I'm connecting from a sipura behind a NAT to an asterisk box behind a different NAT. |
| 10:52.02 | antoniofcano | nobody know about the problem of the two ports :( |
| 10:52.18 | PoWeRKiLL | RoyK when I dial from a sip device to my zap E1 I can hear the ringing audio but I don't want to pass audio until it's answered or busy |
| 10:56.24 | *** join/#asterisk clive- (n=pirch@ndn-165-133-73.telkomadsl.co.za) |
| 11:02.54 | *** join/#asterisk menger (n=menger@dsl-53.69.240.220.rns01-dryb-mel.dsl.comindico.com.au) |
| 11:03.43 | RoyK | PoWeRKiLL: i'm sorry. i don't understand what you mean |
| 11:03.59 | RoyK | PoWeRKiLL: do you want to dial but not pass ringing tone? |
| 11:07.52 | PoWeRKiLL | Yes RoyK |
| 11:08.31 | PoWeRKiLL | RoyK the problem is that the call is consider as answer as soon as it rining |
| 11:09.06 | RoyK | considered answered?? |
| 11:09.14 | RoyK | to where do you call? |
| 11:09.44 | RoyK | are you sure the other side isn't just answering and then playtones(ring)? |
| 11:11.14 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:11.14 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:12.25 | *** join/#asterisk dersteer (n=travis@24-236-197-212.dhcp.aldl.mi.charter.com) |
| 11:14.33 | PoWeRKiLL | I call from a SIP -> * -> IAX2 -> * -> ZAP -> PSTN |
| 11:14.45 | christo | this is weird.. I've just built * on a machine, but when I run asterisk -c I get all sorts of errors and warnings which haven't every trouble me before.. can anybody make sense of this? http://pastebin.ca/24731 |
| 11:15.57 | RoyK | PoWeRKiLL: again, are you sure the one you're calling aren't just answering the call when it first comes in? |
| 11:16.00 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 11:16.00 | RoyK | PoWeRKiLL: pri debug |
| 11:16.04 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 11:16.15 | christo | I think I'll just move the related configs out of /etc/asterisk and retry.. |
| 11:16.54 | *** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
| 11:17.24 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 11:17.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:17.40 | shanky | hi, anyone here that uses asteriskathome, can tell me how to regenerate the extensions.conf file ? |
| 11:20.55 | *** join/#asterisk gvag11 (n=g@84.254.12.236) |
| 11:22.01 | johnm | Anyone have a firm understanting of CLI, ZAP and SIP? Im placing a call which comes in on a Zap channel, and it re-sets the CLI. The call is then carried over SIP, and then goes out over PSTN. I can pass any CLI I like, but if I get an incoming call, and the CLI gets re-set within a specific context, the CLI isn't presented out of the far end correctly. It works for every other context though. ANy thoughts? |
| 11:22.03 | lehel | shanky: make samples (?) |
| 11:22.03 | christo | okay - I have managed to reduce this down to '/usr/lib/asterisk/modules/res_parking.so: cannot open shared object file: No such file or directory' Wouldn't this have been built with the asterisk install?? |
| 11:23.55 | gvag11 | does anyone knows if Asterisk will work with two TE110P on board connected between each other (one master and the other slave) and then start placing calls from one to the other? |
| 11:24.30 | X-Rob | johnm - there's something wrong in your dialplan. There's no difference as far as asterisk is concerned |
| 11:24.52 | X-Rob | christo - put noload=res_parking.so in /etc/asterisk/modules.conf |
| 11:25.26 | johnm | X-Rob: well, every single outgoing call goes out via the same macro. It will re-define CALLERID IF the calleridnum LEN is less than 5. In this case.. I set the CLI so it honours it in the macro. It works from a SIP context, but not from a pstn context. again.. it goes out over the exact same macro. |
| 11:25.29 | X-Rob | gvag11 - yes, it'll work fine. However, you may have issues with channel numbering randomly changing between reboots. It may be better to use a TE4xx card |
| 11:25.55 | *** join/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 11:26.06 | X-Rob | johnm - if you're using 1.2/HEAD (as you should be) it's 'Set(CALLERID(num)=foo) not 'SetCalleridNum(xxx)' |
| 11:26.15 | johnm | X-Rob: thats what Im using. |
| 11:26.24 | Akelavlk | It's possible has ignorepat = 9 and extension _8xxxxxx? |
| 11:26.31 | johnm | and it's Set(CALLERID(number)=foo) :) |
| 11:26.37 | X-Rob | Yeah, that 8) |
| 11:26.42 | X-Rob | Sorry, I'm half pissed. |
| 11:27.15 | X-Rob | tonights photos aren't up yet. |
| 11:27.21 | johnm | X-Rob: new baby? |
| 11:27.25 | X-Rob | yup |
| 11:27.43 | johnm | X-Rob: congrats |
| 11:27.45 | christo | X-Rob - that causes chan_sip.so to fail when it loads - 'undefined symbol: ast_park_call' is the warning issued before it just fails to load |
| 11:27.55 | gvag11 | X-Rob, congratulations .... Bravo |
| 11:28.13 | christo | X-Rob - congrats! :) |
| 11:28.15 | X-Rob | thanks johnm, gvag11 8) |
| 11:28.17 | X-Rob | and christo 8) |
| 11:28.47 | X-Rob | christo - ok. that means something is fubar when loading the module. look at /var/log/asterisk/full, see if you can find out what is missing |
| 11:28.54 | X-Rob | (possibly an 'ldconfig' may fix the problem) |
| 11:28.58 | gvag11 | i want to test the spandsp fax functionality for sending and receiving faxes and i was thinking to save some money by placing the two TE110P boards on the same machine. You think that Asterisk , spandsp and two boards on the same PC will work fine in order to sending and receiving faxes using the boards? |
| 11:29.29 | *** part/#asterisk Akelavlk (n=jansun@82.119.239.141) |
| 11:29.34 | X-Rob | gvag11 - the problem with two TDM cards in one machine is 8000 interrupts per second per card. |
| 11:29.43 | X-Rob | so, yes, two cards will work, it's better to use one. |
| 11:30.07 | PoWeRKiLL | RoyK in the * ZAP side the hangup cause is ok put the sound is directly sent to the sip device so it's answered for him |
| 11:30.55 | gvag11 | X-Rob, you mean 8000 interrupts pers second per card is too much considering performance, right? |
| 11:31.00 | X-Rob | nah |
| 11:31.08 | X-Rob | two is ok, but, it's better to have one |
| 11:53.05 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 11:53.05 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 11:53.39 | johnm | Druken: I should beat you round the head with my hydrospanner for your ignorance! :) |
| 11:54.34 | X-Rob | johnm - yes. It has to listen for the 'beep' fax tones. |
| 11:54.44 | X-Rob | Answer(), Wait(2) |
| 11:54.48 | X-Rob | that usually does it |
| 11:55.35 | johnm | X-Rob: the way it's setup, I'd rather avoid even answering, but I kinda can't avoid that ;) I'll work something out |
| 11:58.48 | Druken | let me guess... don't have a fax number, just using the same number for both? |
| 12:02.19 | *** join/#asterisk aio (n=aio@adsl-61-114-214.sdf.bellsouth.net) |
| 12:02.51 | johnm | not quite. |
| 12:03.05 | johnm | We dont have any fax numbers at all, but we are recieving faxes for other companies mistyping the number |
| 12:03.10 | johnm | it's going to DDI lines |
| 12:03.20 | johnm | I want to intercept them, and re-route them to another outbound line. |
| 12:03.28 | X-Rob | fuck that |
| 12:03.33 | X-Rob | receive the faxes |
| 12:03.33 | johnm | Problem is.. calls are normally passed through. |
| 12:03.37 | X-Rob | sell them to the recipient |
| 12:03.50 | johnm | X-Rob: well.. it's going to another analogue line which is in our building as well. thats got a fax on it |
| 12:04.08 | X-Rob | send 'em a fax, saying 'we have your fax. Call 1-900-xxx-xxx to receive it, xtn [unique fax id]' |
| 12:04.17 | johnm | lol |
| 12:04.43 | Druken | your one sick minded bastard X-Rob, god i love it.. hehe |
| 12:05.14 | johnm | X-Rob: reckon a 1 second wait is long enough? |
| 12:05.19 | X-Rob | you need 2 |
| 12:05.22 | X-Rob | most people don't notice it |
| 12:05.27 | X-Rob | especially in the US |
| 12:05.36 | X-Rob | where it's ring [2 secs] ring [2 secs] |
| 12:05.38 | Druken | pfft.. i reccomend 2 seconds reguardless |
| 12:05.53 | Druken | you ever caleld an IVR over voip that doesn't have a 2 second wait? |
| 12:05.55 | johnm | 2s is just a bit of a pain. long delay before ringtone generates |
| 12:06.02 | X-Rob | in au it's ring[.75][.25]ring[.75][1.5]..repeat |
| 12:06.03 | johnm | Druken: yep. |
| 12:06.31 | X-Rob | anyway |
| 12:06.47 | X-Rob | wifey says I have to come upstairs and spend time with jade |
| 12:06.57 | X-Rob | http://aussievoip.com/jade for those that haven't seen her yet. |
| 12:07.03 | Druken | go spend time with the kids! :) |
| 12:09.02 | *** join/#asterisk KeX_WorX (n=chris@83-65-129-46.paris-lodron.xdsl-line.inode.at) |
| 12:09.05 | KeX_WorX | hi |
| 12:09.27 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:09.55 | Druken | welcome to the idlers club |
| 12:10.03 | KeX_WorX | : ) |
| 12:10.39 | Delvar | im not idle i just dont like talking to you lot |
| 12:11.10 | Druken | well that's ok, i don't ever remember talking to you... so we're good :) |
| 12:12.47 | *** join/#asterisk szer (n=Miranda@217.116.36.22) |
| 12:12.57 | szer | hi all |
| 12:14.30 | aio | ok - got a really dumb question and asterisk may be overkill, but I am thinking of doing podcasting and it looks to be a pain to record both ends of the conversation with skype, so i was thinking of setting up asterisk and having callers dial in |
| 12:14.38 | aio | is this reasonable? reasonably easy? |
| 12:14.56 | johnm | aio: yes, and check out the Monitor app. |
| 12:15.12 | johnm | aio: you will probably want ot use sox or something to trasncode the audio at the end into a more suitable format. |
| 12:16.39 | aio | johnm would it be reasonable to record from my client? or better to record from the server? |
| 12:16.42 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 12:17.04 | johnm | aio: the server ideally. |
| 12:17.14 | johnm | aio: depends if you are using asterisk as your client or not. |
| 12:17.46 | aio | johnm haven't really planned that out. i was thinking of setting up the server and just seeing which client i like best. |
| 12:17.56 | szer | i've got some quality problem with chanspy when i try to spy on calls using iax. Anyone knows the cause of it? |
| 12:18.11 | johnm | aio: I like sjphone ;) |
| 12:21.54 | szer | <PROTECTED> |
| 12:28.24 | *** part/#asterisk samyl (n=samyl@194.167.18.244) |
| 12:28.48 | *** join/#asterisk azrishahril (n=azrishah@60.50.196.59) |
| 12:30.18 | christo | I'm gonna try the latest stable * and hope that there aren't any missing modules this time ;) |
| 12:31.24 | KeX_WorX | hi |
| 12:31.39 | KeX_WorX | anyone using asterisk 1.2 beta ? |
| 12:32.42 | KeX_WorX | i've a question. when two phones talk to each other and 1 call one of this phones, i get dialstatus "Unavailable" |
| 12:32.46 | KeX_WorX | why not busy ? |
| 12:32.59 | Attila_Kovacs | Do ZAPHFC + WCFXO work together? |
| 12:33.10 | aio | ok - i just got asterisk installed and running on ubuntu. i installed the asterisk-gtk-console package, but i don't see a binary to start it. anybody know? |
| 12:33.19 | KeX_WorX | and i can hear the ringing tone on the called phone (in the background) |
| 12:35.12 | gvag11 | is there somebody that has already setup Asterisk with Spandsp for faxing? Does it work fine? |
| 13:13.30 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 13:13.30 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 13:13.30 | mut | pastebin ya config |
| 13:15.05 | *** join/#asterisk AsterNov (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk) |
| 13:15.59 | *** join/#asterisk Dio_ (n=dima@arkadia.soborka.net) |
| 13:16.36 | Dio_ | hello, is there any bug marshals who can help with re-openning a bug? |
| 13:17.23 | cpatry | which one? |
| 13:17.50 | Dio_ | 4973 |
| 13:18.09 | Dio_ | cpatry: I would like to add comments there |
| 13:18.45 | cpatry | go ahead. |
| 13:19.24 | Katty | mew. |
| 13:20.43 | Dio_ | cpatry: thx. adding now.. |
| 13:20.50 | Katty | mew? |
| 13:21.18 | lathos42 | Morning Katty |
| 13:24.10 | *** join/#asterisk zoo (i=nobody@ip-98-16.travedsl.de) |
| 13:24.12 | zoo | hello |
| 13:24.34 | Katty | lathos42: mew. |
| 13:24.35 | zoo | I am trying to get music-on-hold to work. I only configured " |
| 13:24.40 | zoo | default => quietmp3:/var/lib/asterisk/mohmp3 |
| 13:24.48 | zoo | " but it does not work :( |
| 13:25.03 | fishboy1669 | hi |
| 13:25.11 | zoo | * says "Started music on hold, class 'default'" but i dont hear a thing :( |
| 13:25.24 | fishboy1669 | anyone here ever used ChanIsAvail with sip channels? |
| 13:25.32 | zoo | in that directory there are three mp3-files and mpg123 is in path |
| 13:25.36 | zoo | any hint? |
| 13:26.12 | Katty | zoo: the only thing i can think of is maybe you don't have the codec enabled. |
| 13:26.21 | Katty | zoo: like it's not allowed or something. |
| 13:26.31 | zoo | Katty: codec mp3? |
| 13:26.43 | Katty | zoo: that's what i said, bunny bread. |
| 13:27.58 | zoo | but dialling a test exten which has ,1,WaitMusicOnHold(default) works |
| 13:28.10 | Katty | i gave you my advice. |
| 13:28.15 | Katty | you don't have to take it. |
| 13:28.28 | *** join/#asterisk Hmmhesays (n=Neg@24-117-213-113.cpe.cableone.net) |
| 13:28.37 | file[laptop] | Hmmhesays! |
| 13:29.13 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:29.50 | zoo | Katty: okay, sorry. i will try to take it |
| 13:29.51 | iDunno | fishboy1669: yes, I'm using ChanIsAvail. |
| 13:30.44 | iDunno | fishboy1669: it doesn't tell you if the channel is busy, though... for that you need to use the other thing, quite documented somewhere, possibly on voip-info.org |
| 13:30.55 | Katty | zoo: no one ever said i was right though. |
| 13:31.13 | Katty | Hmmhesays: sleepy. |
| 13:31.39 | file[laptop] | I'm without power right now :( |
| 13:31.55 | Katty | :<< |
| 13:32.41 | zoo | Katty: but allow=mp3 is not an allowed value :( |
| 13:33.32 | Ariel_ | morning all |
| 13:33.49 | PoWeRKiLL | Hi Ariel_ :) |
| 13:34.26 | file[laptop] | my primary router and ADSL are on a UPS |
| 13:34.35 | tzanger | he's got a hand-crank laptop |
| 13:34.38 | iDunno | cunning! |
| 13:35.15 | *** part/#asterisk toddf (n=toddf@ns0.fries.net) |
| 13:35.49 | *** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 13:36.02 | tzanger | yes, he's a cunning linguist. |
| 13:36.15 | tzanger | we must crush the resistance before the rainy seasons! |
| 13:36.30 | shido6 | anyone have a mac? |
| 13:36.57 | jake1932 | shido6: join asterisk-mac |
| 13:37.11 | lathos42 | shido6: I've got an iBook, but I dont have * running on it |
| 13:38.36 | iDunno | shido6: lots of people have macs, apparently apple sell them at extorsionate prices and people buy them. I don't understand why. |
| 13:38.57 | shido6 | my adobe creative suite .cdr wont mount in disk utility |
| 13:39.11 | Ariel_ | PoWeRKiLL, hello |
| 13:39.12 | shido6 | im on an intel box, iDunno. |
| 13:39.14 | shido6 | :) |
| 13:39.25 | zoo | Katty: i solved it, i had a global canreinvite=yes which prevented MOH, because * was not in the media stream any more :( |
| 13:39.54 | Bonzai070 | i read somewere that mac is going to use intel cpu's some time soon |
| 13:39.55 | iDunno | the machine that's running irssi is an amd64, the machine that I'm sitting at at work is a P4, and the one just to the left where the asterisk install is is an amd64 :) |
| 13:40.18 | Ariel_ | mac don't run the upgrade program for sipura firmware.... but i like mac's |
| 13:40.30 | shido6 | Bonzai070 it works today. |
| 13:40.34 | shido6 | Im running it. |
| 13:40.40 | fishboy1669 | iDunno how r u using ChanIsAvail? |
| 13:40.44 | fishboy1669 | and whats the other thingg |
| 13:41.23 | shido6 | file, you have toast? |
| 13:41.33 | shido6 | toast wont run on intel :( |
| 13:42.30 | file[laptop] | it's titanium flavor |
| 13:42.37 | PakiPenguin | shido6, you running os x on intel? |
| 13:42.41 | shido6 | yeah |
| 13:42.43 | fishboy1669 | IDonno u still there? |
| 13:42.48 | jake1932 | this says canreinvite "won't be pretty" behind NAT http://www.voip-info.org/wiki-Asterisk+sip+canreinvite - but it seems to work consistently for me |
| 13:43.12 | Ariel_ | jake1932, canreinvite=yes |
| 13:43.18 | shido6 | whats your net speed, file? |
| 13:43.35 | file[laptop] | shido6: well all that stuff is sorta unavailable, it's on my workstation that has no power :P |
| 13:43.37 | psycodad_ | Can somebody explain why I cant get the FXO port to work on my TDM13B.. ztcfg -v report all 4 channels as working but asterisk says " Unable to register channel '1'" |
| 13:43.47 | psycodad_ | Could that be due to shared IRQs ? |
| 13:43.52 | jake1932 | Ariel_: i'm saying it does work |
| 13:44.17 | iDunno | fishboy1669: with ChanIsAvail(SIP/phone1000&SIP/phone1001&...&SIP/phone1007) |
| 13:44.29 | iDunno | fishboy1669: and just to check which phones are actually registered. |
| 13:44.41 | Ariel_ | jake1932, in some systems which are configured correctly it works. But it's a problem for most to get working correctly. |
| 13:44.48 | fishboy1669 | aha i see |
| 13:44.50 | jake1932 | ok |
| 13:44.51 | *** part/#asterisk criptos (n=criptos@201.145.229.189) |
| 13:45.00 | fishboy1669 | and how do i find which ones are engaged? |
| 13:45.20 | iDunno | fishboy1669: at the moment I don't take in to account that they might be in use, I need to write some more rules in the extensions to take in to account that people might be on the phone, though it looks like the hardware phones report back a busy when they're busy :) |
| 13:45.29 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-68-255-63-230.dsl.lgtpmi.ameritech.net) |
| 13:45.30 | zoo | can i set custom variables for each phone, that i can pick up in a dialplan? i would like to use a phone-depending ${AREACODE}. Is that possible? |
| 13:45.50 | tzanger | zoo: you could do it with GetDB |
| 13:45.50 | iDunno | fishboy1669: you use Groups, IIRC, and increment it when a call comes in. I haven't done that bit yet ;) |
| 13:45.59 | tzanger | er DBGet |
| 13:45.59 | fishboy1669 | the issue i have is the phones i am using ip2006 and grandstream have multiple lines |
| 13:45.59 | tzanger | just look up the extension |
| 13:46.10 | fishboy1669 | but i want asterisk to think there engaged if one line is used |
| 13:46.11 | nfi|ermes | i have asteriisk 1.2 installed in my so |
| 13:46.26 | fishboy1669 | i cant kill the extra lines on the phones cos they are used for call transferes |
| 13:46.29 | nfi|ermes | if now i compile asterisk 1.0.0 is a problem ? |
| 13:46.34 | nfi|ermes | if now i compile asterisk 1.0.9 is a problem ? |
| 13:46.48 | christo | I have a totally unexplainable problem - I have built two versions on asterisk here, but both times I don't get a res_parking.so module, so I'm unable to start the server. What could be happening? |
| 13:47.09 | file[laptop] | christo: res_parking is deprecated, it's res_features now |
| 13:47.10 | szer | i need some help with chanspy. when i spying on iax calls or try to spy with an iax phone i've got discontinous voice |
| 13:47.25 | szer | i havent got any idea why is this |
| 13:48.11 | nfi|ermes | can i use asterisk 1.2 with zaptel 1.0.9 , zaphfc and libpri 1.0.9 ? |
| 13:49.16 | christo | flie[laptop] but when I start asterisk, it says /usr/lib/asterisk/modules/res_parking.so: cannot open shared object file - So should I symlink res_features.so over to res_parking.so just to keep it happy? |
| 13:49.40 | file[laptop] | christo: update your modules.conf |
| 13:49.46 | christo | oh |
| 13:50.06 | shido6 | I like ircle and coloquy |
| 13:50.26 | file[laptop] | I use Colloquy, but it dislikes #asterisk |
| 13:50.49 | shido6 | hrmm |
| 13:50.53 | shido6 | do you use macsql 3.3, file? |
| 13:51.01 | file[laptop] | no |
| 13:51.03 | Katty | my silly self wants to nap forever |
| 13:51.12 | shido6 | bbedit? |
| 13:51.22 | file[laptop] | nope |
| 13:51.33 | christo | file[laptop] well that worked.. thanks! I've been trying to figure that out for about 2 hours. Should it be stuffed into the wiki someplace perhaps? |
| 13:51.44 | christo | under 'build gotchas' or something |
| 13:51.48 | file[laptop] | christo: that change happened a LONG time ago... |
| 13:52.00 | christo | oh |
| 13:52.05 | christo | where was it documented? |
| 13:52.11 | file[laptop] | I don't even remember |
| 13:52.28 | razu | can anyone tell me, what does this mean : pbx_spool.c:229 attempt_thread: Call failed to go through, reason 5 |
| 13:52.30 | razu | ? |
| 13:52.32 | christo | I admit I'm not a * guru, but I figured I was doing all the 'normal' sensible things when building these bits... |
| 13:52.36 | christo | ..nevermind :) |
| 13:52.55 | file[laptop] | christo: you gotta think though, "hrm... maybe my config files are old!" |
| 13:53.12 | file[laptop] | but meh |
| 13:53.16 | iDunno | heh |
| 13:53.24 | iDunno | if it all breaks, it's the fault of the config ;) |
| 13:53.38 | christo | file[laptop] - I did and I moved them out of /etc and did a 'make samples' |
| 13:53.44 | Bonzai070 | lol blame it on the config lol |
| 13:54.43 | christo | but there's no way I would have known that res_parking.so had been renamed to res_features.so unless it was mentioned someplace.. well I guess if I was always plugged into *-dev or *-users, then perhaps I might have picked that up... |
| 13:54.49 | christo | anyway, no harm done :) |
| 13:56.33 | iDunno | it's mentioned lots. |
| 13:56.33 | drbrown_ | is there a way to detect an extension that a call is being transfered from IE a variable???? |
| 13:57.46 | iDunno | (or at least, I've seen it mentioned lots in the wiki, but now I can't find a reference ;) |
| 13:57.50 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 13:58.05 | *** join/#asterisk WorkTooMuch (n=work@82.148.188.56) |
| 13:58.33 | WorkTooMuch | Hello has anyone had any sucsess with SpanDSP? |
| 13:59.05 | *** join/#asterisk fordvoice (n=chrisf0r@rrcs-70-61-133-91.central.biz.rr.com) |
| 14:01.35 | *** join/#asterisk The_Duke (n=The_Duke@80.92.64.103) |
| 14:01.40 | The_Duke | Hello |
| 14:01.41 | christo | WorlTooMuch - yes I have |
| 14:02.32 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 14:03.39 | WorkTooMuch | christo, is it stable? |
| 14:03.59 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:04.23 | The_Duke | does someone know if I can add Sip Users with a static IP and Port, which do not support SIP REGISTER (e.g. Cisco Call Manager, SER) to Realtime SIP??? |
| 14:05.41 | *** join/#asterisk Defraz (n=t0tal@72.24.26.215) |
| 14:05.44 | *** join/#asterisk marv[work] (n=timr@border0hsv.asterisksgi.com) |
| 14:07.27 | *** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com) |
| 14:07.48 | protien | i cant work out why i cant hear any calls i get from outside network |
| 14:07.54 | protien | ive set up asterisk with a network dialer, im behind a nat |
| 14:08.00 | protien | ive got to the stage i can get incoming calls |
| 14:08.05 | protien | but when i do the echo test |
| 14:08.08 | protien | i cant hear anything |
| 14:08.43 | file[laptop] | do you have externip and localnet set if you're using SIP? |
| 14:08.55 | protien | i have neither set |
| 14:09.18 | christo | WorkTooMuch - yes, but I suffered some slip when faxing simultaneously over 20 or so channels, so I had to write a manager script to keep checking on how many channels were in use before firing off new ones |
| 14:10.28 | christo | that was using a perl script to create call files carrying the txfax command |
| 14:11.09 | bendy24 | YAY |
| 14:11.11 | bendy24 | ITEM(S) SHIPPED: |
| 14:11.15 | bendy24 | 1. Asterisk: The Future of Telephony |
| 14:11.53 | protien | im actually using iax file[laptop] |
| 14:12.03 | WorkTooMuch | christo, When I try to send fax asterisk it starts with hig speed and then it just get lower and lower til the conection is lost. |
| 14:12.08 | WorkTooMuch | :( |
| 14:12.40 | shido6 | what codec are you using WorkTooMuc? |
| 14:12.56 | Katty | file[laptop]: find a nap for me. |
| 14:13.04 | file[laptop] | Katty: have you tried eBay? |
| 14:13.15 | WorkTooMuch | well It shuld be u-law or a-law but I have to retest it |
| 14:13.17 | Katty | file[laptop]: yes. |
| 14:13.37 | Katty | file[laptop]: i didn't have the time it required :< |
| 14:14.55 | protien | hm |
| 14:15.01 | *** join/#asterisk froud (n=froud@ndn-165-134-136.telkomadsl.co.za) |
| 14:15.15 | file[laptop] | Katty: darn |
| 14:15.42 | WorkTooMuch | brb the admin has to take down the firewall for 15 min :( |
| 14:17.55 | *** join/#asterisk shanky (n=shanky@238.Red-80-33-29.staticIP.rima-tde.net) |
| 14:18.00 | shanky | hi again |
| 14:18.31 | newl | Management finally accepted the fact that no matter what they did, their SOE desktops were going to keep getting infected and the firewall wasn't helping matters any. ;) |
| 14:20.42 | shanky | I have a problem with the incoming calls |
| 14:21.06 | shanky | I have used google but I can't find any tip |
| 14:21.07 | tzanger | stop giving out your # then |
| 14:21.29 | shanky | I get this from the asterisk cli: |
| 14:21.31 | shanky | <PROTECTED> |
| 14:21.49 | Bonzai070 | question i have a pri why will it drop calls when the 28th call comes in?.. |
| 14:24.32 | Bonzai070 | like i was talking to clive- and some one else called in and blam my call got a long angaged tone |
| 14:25.56 | protien | im using sip, got 1way audio with freeworlddialup connection (ie remote connections), im also behind a nat |
| 14:26.01 | protien | anyone able to help me work through this problem |
| 14:26.07 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 14:26.18 | *** join/#asterisk ACiDV (n=acidvici@26-121.dr.cgocable.ca) |
| 14:28.01 | *** join/#asterisk crash3m (i=crash3m@unaffiliated/crash3m) |
| 14:28.08 | ACiDV | I have a Cisco 1760V (SIP) gateway that is connected to Asterisk (1.0.9), for all incoming calls, Asterisk try to match an extension in context default. but in my [gateway] i have set a context=gateway-incoming. Why Asterisk try to match in [general]context=xxx and not [device]context=xxx ? |
| 14:29.31 | *** join/#asterisk Weezey (n=ohno@206.210.109.226) |
| 14:29.31 | BrianR___ | has anyone here used ADSI over an ATA? |
| 14:29.37 | *** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
| 14:34.09 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 14:34.09 | *** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now! |
| 14:36.07 | loud | is it normal for iLBC to consume 28/30 kbps on a satellite link ? |
| 14:36.21 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 14:36.37 | clive- | loud sounds like too litlle imho |
| 14:37.34 | christo | file[laptop] did codec_g723_1.so get depracated too? |
| 14:37.39 | mut | anyone here know of a good ftpd to do virtual users? |
| 14:37.44 | mut | other than proftpd |
| 14:38.04 | christo | mut - I've heard vsftp is good |
| 14:39.36 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 14:40.27 | *** join/#asterisk tdonahue (n=tdonahue@64.201.13.50) |
| 14:40.39 | tdonahue | good morning all |
| 14:40.53 | nfi|ermes | asterisk: relocation error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: ast_config_load |
| 14:41.22 | ManxPower | ACiDV, Asterisk doesn't think the incoming call from the Cisco matches the [device] section |
| 14:41.25 | nfi|ermes | anyone can help me to find out the problem .....and the solution ...? |
| 14:41.34 | file[laptop] | nfi|ermes: did you downgrade from CVS to stable? |
| 14:41.52 | tdonahue | how can I get asterisk to show me the context that a call is coming in from in the console? i have a call that is not ending up in the expected place |
| 14:41.54 | file[laptop] | christo: codec_g723_1.so never came with Asterisk |
| 14:41.56 | nfi|ermes | i had 1.2....the i built 1.0.9 |
| 14:42.06 | *** join/#asterisk gabb0 (n=gabb0@131.202.90.23) |
| 14:42.09 | file[laptop] | nfi|ermes: you need to wipe your modules directory for Asterisk, and then do a make install again |
| 14:42.15 | gabb0 | hello all |
| 14:42.15 | file[laptop] | nfi|ermes: because stable does not have res_config_mysql |
| 14:42.16 | *** join/#asterisk preeezy (i=jjmtrev@203-173-26-187.dyn.iinet.net.au) |
| 14:42.24 | ManxPower | nfi|ermes, and you didn't read the BIG BANNER when you did a "make install" for 1.0.9 that talks about downgrading, did you? |
| 14:42.32 | preeezy | Oct 7 00:08:00 WARNING[20804]: channel.c:2052 set_format: Unable to find a codec translation path from gsm to g729 |
| 14:42.35 | file[laptop] | nobody ever does |
| 14:42.41 | gabb0 | quick question about meetme for someone familiar with it. anyone around |
| 14:42.42 | preeezy | im getting this error continually on connections to fwd network |
| 14:42.45 | file[laptop] | preeezy: you don't have the g729 codec installed, so Asterisk can't transcode |
| 14:42.46 | preeezy | can i work aorund it? |
| 14:42.49 | preeezy | ah |
| 14:42.51 | nfi|ermes | i didn t |
| 14:42.55 | preeezy | how can i obtain g729? |
| 14:42.59 | ManxPower | preeezy, either but a g729 license or don't allow G729 for that device |
| 14:43.02 | file[laptop] | preeezy: pay $10 for channel to Digium |
| 14:43.03 | Delvar | <preeezy>: dont use g729! or buy licance from digium |
| 14:43.05 | file[laptop] | er per channel |
| 14:43.08 | ManxPower | but == buy |
| 14:43.32 | christo | file[laptop] - okay thanks. I've removed that from my modules.conf and everything seems to be okay now.. |
| 14:43.32 | preeezy | hm whats so good about g729 that they charge for it |
| 14:43.35 | Delvar | ill buy buts |
| 14:43.42 | file[laptop] | preeezy: it's licensed |
| 14:43.48 | file[laptop] | it's not a free codec. |
| 14:43.55 | *** join/#asterisk SwK[Work] (n=SwK@border0hsv.asterisksgi.com) |
| 14:44.15 | ManxPower | The patent holders for G729 charge to use it. |
| 14:44.20 | preeezy | yeah i know what you mean but |
| 14:44.25 | ManxPower | Digium just passes on the licensing fees. |
| 14:44.29 | preeezy | is it really special? |
| 14:44.33 | preeezy | or what |
| 14:44.44 | file[laptop] | least bandwidth and decent audio quality. |
| 14:44.51 | ManxPower | preeezy, for a long time G729 and G723.1 were the only low bandsidth codecs. |
| 14:45.02 | preeezy | gsm is the primary free one now? |
| 14:45.18 | *** join/#asterisk Goshen (n=Goshen@c-67-172-238-57.hsd1.ut.comcast.net) |
| 14:45.29 | ManxPower | preeezy, there are many of them now. |
| 14:45.48 | gabb0 | how does * guarentee that in a meetme conf, if person A presses 1 that it doesn't also get detected as person B pressing 1? |
| 14:46.12 | bkw_ | dtmf is squelched |
| 14:46.17 | *** join/#asterisk WorkTooMuch (n=work@82.148.188.56) |
| 14:46.37 | gabb0 | squelched meaning? |
| 14:47.05 | johnm | blocked. |
| 14:47.21 | gabb0 | but the audio portion still goes through |
| 14:47.37 | gabb0 | does it now? |
| 14:47.41 | gabb0 | does it not? |
| 14:47.53 | zoo | which variable holds the phone i am using? |
| 14:47.54 | BrianR___ | there's now some pretty decent alternatives like ilbc... |
| 14:49.42 | *** part/#asterisk Dio_ (n=dima@arkadia.soborka.net) |
| 14:50.01 | gabb0 | bkw_, johnm, how does asterisk or better yet, where does asterisk stop the dtmf tone from being played to the entire conference participants. is it within the meetme code or within the chan_zap code |
| 14:50.22 | *** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
| 14:50.41 | Hmmhesays | probably in meetme |
| 14:51.28 | file[laptop] | well DTMF doesn't travel as sound or whatever, unless you are using inband and don't have dtmfmode set for SIP (doesn't fire up the DSP) |
| 14:51.47 | gabb0 | I'm just using PRIs |
| 14:51.50 | gabb0 | at the moment |
| 14:51.51 | johnm | gabb0: I have no idea im afraid. |
| 14:52.00 | Katty | Hmmhesays: we are number 13 in rank amongst 114 stores. |
| 14:52.03 | file[laptop] | it's recognized in the channel driver or audio medium (like RTP or chan_zap or chan_sip) and sent as a DTMF frame |
| 14:52.08 | Katty | Hmmhesays: isn't that neat. |
| 14:52.15 | file[laptop] | the application, or other channel, then does whatever it wants with it |
| 14:53.09 | Katty | Hmmhesays: nextel dealers, that is. |
| 14:53.17 | gabb0 | so the "if" statements in meetme handle the dtmf and implement whatever feature and then after the "feature" has been completed it sends writes the dtmf out (plays the audio) |
| 14:53.19 | gabb0 | ? |
| 14:53.38 | file[laptop] | gabb0: but meetme doesn't allow DTMF to go usually, it blocks it |
| 14:54.58 | gabb0 | has it always done that? because I know quite some time ago it didn't. Just wondering if this was something newly added within that last year or maybe even more. |
| 14:55.23 | gabb0 | it's been a while since I looked through it any great detail. I know lots has changed. |
| 14:55.47 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 14:56.11 | file[laptop] | as far as I can remember |
| 14:56.26 | gabb0 | does conf_flush do this |
| 14:56.41 | file[laptop] | no... |
| 14:57.15 | file[laptop] | DTMF comes in as frames, look for AST_FRAME_DTMF |
| 14:58.46 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 14:59.15 | gabb0 | yeah but conf_flush gets called after the dtmf frame is handled |
| 14:59.19 | gabb0 | that is why i asked |
| 14:59.29 | *** join/#asterisk iCEBrkr (i=icebrkr@24.129.130.158) |
| 15:00.07 | *** join/#asterisk luke-jr_ (n=luke-jr@CPE-65-26-133-171.kc.res.rr.com) |
| 15:00.59 | file[laptop] | DTMF never comes in as audio |
| 15:01.04 | file[laptop] | it's intercepted ahead of time |
| 15:01.14 | file[laptop] | UNLESS, it's not intercepted by the channel driver under some circumstances |
| 15:02.06 | tdonahue | how can I get asterisk to show me the context that a call is coming in from in the console? i have a call that is not ending up in the expected place |
| 15:03.05 | gabb0 | file[laptop], I've seen dtmf get handled but the "audio" from the dtmf key is heard by all |
| 15:03.20 | gabb0 | that is what is confusing me |
| 15:03.28 | file[laptop] | on a PRI? |
| 15:03.51 | gabb0 | yes |
| 15:03.52 | file[laptop] | I could see how that could happen with SIP, but I'm not a PRI person |
| 15:04.04 | gabb0 | oh |
| 15:04.18 | *** join/#asterisk funxion (n=nunya@mtnuser.icgws.com) |
| 15:04.22 | file[laptop] | on SIP the device could send DTMF as both out of band and inband at the same time, I've seen something that did that |
| 15:04.23 | *** join/#asterisk Sk3tCh (n=wrickout@86.127.41.114) |
| 15:05.01 | gabb0 | well, this would be inband |
| 15:05.14 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:06.00 | Sk3tCh | hhi |
| 15:07.21 | *** join/#asterisk gvag11 (n=g@ppp22-adsl-105.ath.forthnet.gr) |
| 15:08.11 | iDunno | sup Katty? |
| 15:08.27 | Katty | ceiling |
| 15:08.44 | zoo | How do i do this "database get nebenstelle/11900 onkz" as DBget()? I am not managing to do that |
| 15:09.35 | zoo | DBget(ONKZ=/nebenstelle/${CALLERIDNUM}/onkz) does not detect the family |
| 15:09.55 | blitzrage | what the HELL is Level3 and Cogent doing?! |
| 15:10.10 | file[laptop] | being a pain in my ass |
| 15:10.28 | blitzrage | mine too |
| 15:10.40 | mogorman | lol seem to be everyones problem today |
| 15:10.41 | file[laptop] | welcome to the club, have you paid your fees? |
| 15:11.08 | Katty | it obviously needs more hugs. |
| 15:11.13 | mut | heh |
| 15:11.29 | blitzrage | file[laptop]: I think so... |
| 15:11.36 | mut | i seen 100 bajillion people today all at once go OMG WoW DIED!? |
| 15:11.41 | blitzrage | I have things that can route one way, but not the other |
| 15:11.47 | mut | cause of l3 'n cognt |
| 15:11.51 | blitzrage | mut: lol |
| 15:11.58 | blitzrage | not WoW!!!!!???!?!! |
| 15:12.07 | mut | i had to laugh |
| 15:12.12 | mut | cuase it was in like 10 chans i was in |
| 15:12.16 | mut | 4 diff networks |
| 15:12.59 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 15:13.52 | blitzrage | heh :) |
| 15:14.12 | file[laptop] | blitzrage: aren't you supposed to be getting ready and stuff? |
| 15:14.19 | blitzrage | file[laptop]: yes!!!!!!! |
| 15:14.30 | blitzrage | file[laptop]: emergency networking problem too though... trying to do btoh |
| 15:14.31 | file[laptop] | WELL HOP TO IT! |
| 15:14.31 | blitzrage | both* |
| 15:15.09 | Weezey | blitzrage: http://ask.slashdot.org/article.pl?sid=05/10/05/2247207&tid=95&tid=187&tid=4 |
| 15:15.13 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.231) |
| 15:15.24 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 15:15.30 | blitzrage | file[laptop]: ummmm... yah... the whole emergency routing thing and Level(3) and Cogent FUCKING ME |
| 15:15.46 | blitzrage | who can I call and scream at? |
| 15:16.01 | file[laptop] | not me!!! |
| 15:16.05 | Katty | blitzrage: the mirror |
| 15:16.31 | Ariel_ | ok so what is the command to ignore someone on the channel |
| 15:16.35 | Katty | blitzrage: i'm sure you just screaming at someone is keeping them from doing whatever it is that needs to be done to fix it. |
| 15:16.44 | Katty | Ariel_: i think it's /ignore $nick |
| 15:17.32 | azrishahril | does anyone here use oh323 ? |
| 15:18.16 | blitzrage | Katty: possible :) |
| 15:18.33 | Ariel_ | Katty, yes your correct thanks |
| 15:18.42 | Katty | Ariel_: yay! |
| 15:19.58 | odie_flocon | Hey Katty |
| 15:20.02 | odie_flocon | Hey Ariel_ |
| 15:20.12 | asterisk99 | Is anyone here using a TE110P (single-span T1/E1) card or the 2-3-4 span variants? i have PRI configuration problems |
| 15:20.21 | Katty | odie_flocon: hihi |
| 15:20.22 | odie_flocon | hey blitz |
| 15:21.07 | KranZ | moops |
| 15:21.38 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:21.41 | ian_k | asterisk99: sup? |
| 15:22.05 | Katty | bkw_: where has darthclue been? |
| 15:22.12 | asterisk99 | ian_k: Msg; ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 15:22.41 | asterisk99 | ian_k: that error from ztcfg |
| 15:22.46 | ian_k | asterisk99: make sure /etc/zaptel.conf is setup correctly |
| 15:24.01 | KranZ | asterisk99: check dmesg where the driver loads |
| 15:24.04 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 15:24.11 | KranZ | looks like your driver didnt compile correctly |
| 15:24.13 | *** join/#asterisk toddf (n=toddf@adsl-65-70-118-15.dsl.okcyok.swbell.net) |
| 15:24.15 | Ariel_ | Is there someone here that can translate this into english more me. Or better yet into tech talk for me? Please: http://pastebin.ca/24734 |
| 15:24.43 | Ariel_ | more/ for |
| 15:25.15 | *** join/#asterisk _santiago_ (n=santiago@208.195.215.231) |
| 15:25.24 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166094161.nb.aliant.net) |
| 15:25.55 | Inv_arp | Ariel_: seems like he cant call out thru his sip device |
| 15:26.09 | asterisk99 | KranZ: no error messages in dmesg that I can see |
| 15:26.41 | Ariel_ | Inv_arp, yes but which from pots line to sip or sip to pots line? |
| 15:26.41 | christo | Ariel_ it's pretty bad english, but it looks like you need to add a _9. extension to your dial plan to allow outside lines... or tell him that you don't offer that.. or something |
| 15:27.01 | asterisk99 | KranZ: ian_k: span=1,1,0,esf,b8zs |
| 15:27.19 | asterisk99 | KranZ: ian_k: bchan=1-23 |
| 15:27.27 | Inv_arp | bet its sip to pots |
| 15:27.31 | ian_k | aterisk99: are you providing timing to the telco? |
| 15:27.32 | asterisk99 | KranZ: ian_k: dchann=24 |
| 15:27.45 | asterisk99 | KranZ: ian_k: dchan=24 (correction) |
| 15:28.06 | asterisk99 | ian_k: timimg to telco??? where do I define that? |
| 15:29.13 | ian_k | aterisk99: maybe it is span=1,0,0,esf,b8zs |
| 15:29.27 | ian_k | aterisk99: instead of span=1,1,0,esf,b8zs |
| 15:29.44 | *** join/#asterisk ZX81 (n=ZX81@222-153-100-242.jetstream.xtra.co.nz) |
| 15:29.46 | ZX81 | hihi |
| 15:29.51 | file[laptop] | uh oh Matt |
| 15:29.52 | blitzrage | yo |
| 15:29.54 | ZX81 | :) |
| 15:29.56 | ZX81 | heh |
| 15:30.02 | ZX81 | <-- has oh323 problem |
| 15:30.19 | blitzrage | <-- has Level(3) / Cogent problems |
| 15:30.22 | ZX81 | even worse than having to use openh323 in the first place |
| 15:30.24 | ZX81 | :) |
| 15:30.39 | ZX81 | lol |
| 15:30.50 | Katty | yay for not making a mess! |
| 15:31.00 | wunderkin | blitzrage: i heard level3 stopped peering with all other networks yesterday |
| 15:31.17 | KranZ | asterisk99: it's prolly 1,1,0 |
| 15:31.27 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 15:31.28 | ZX81 | which is the best 323 channel to use? |
| 15:31.28 | KranZ | but the pri will come up regardless |
| 15:31.40 | blitzrage | wunderkin: yep... they did -- I'm feeling the effects today |
| 15:31.41 | ZX81 | trying the inaccess one with 1.2beta |
| 15:31.43 | ZX81 | :) |
| 15:31.47 | wunderkin | suckage |
| 15:31.51 | KranZ | asterisk99: did you run ztcfg -vv? |
| 15:31.52 | asterisk99 | KranZ: The LED on the card is off |
| 15:32.00 | KranZ | means the driver is loaded |
| 15:32.02 | ZX81 | but I get an unsatified link error |
| 15:32.03 | ian_k | asterisk99: your module is not loaded |
| 15:32.08 | KranZ | heh |
| 15:32.11 | asterisk99 | KranZ: Yes, ztcfg gave me that error |
| 15:32.16 | wunderkin | network nazis |
| 15:32.34 | wunderkin | we super we make our own intraweb |
| 15:32.39 | asterisk99 | wunderkin: No network for you!!!!!!!!! |
| 15:32.53 | ZX81 | :) |
| 15:33.02 | asterisk99 | wunderkin: variation of No soup for you!!!!!!!!! |
| 15:33.28 | asterisk99 | ian_k: I thought I loaded it by a modprobe zaptel |
| 15:33.54 | KranZ | asterisk99: msg me your dmesg |
| 15:34.25 | file[laptop] | I have not consumed food yet today |
| 15:34.26 | KranZ | asterisk99: you need to modprobe the driver for the card |
| 15:34.34 | KranZ | and that will also load zaptel |
| 15:34.46 | Katty | file[laptop]: go eat. |
| 15:34.46 | asterisk99 | ian_k: lsmod shows zaptel loaded |
| 15:34.53 | KranZ | you should have two |
| 15:34.56 | asterisk99 | KranZ: lsmod shows zaptel loaded |
| 15:34.58 | KranZ | i dont know the one for single span |
| 15:35.11 | KranZ | but i do "modprobe wct4xxp" and it loads that with zaptel |
| 15:35.24 | KranZ | so find out what your module name is and load it |
| 15:36.47 | asterisk99 | KranZ: I have the sigle span T1/E1... |
| 15:38.00 | KranZ | do a "modprobe wct1xxp" |
| 15:39.38 | uter | is there anybody who has a snomphone with working, blinking LEDs? |
| 15:39.41 | malcolmd | single span t1/e1 == te110p == wcte11xp; wct1xxp == the old t100p/e100p cards |
| 15:39.57 | *** join/#asterisk ^X-works (n=r0x0r@81-208-62-98.ip.fastwebnet.it) |
| 15:40.04 | uter | on my phone the don't want to blink |
| 15:40.04 | KranZ | oh |
| 15:40.38 | uter | i think it's a problem with the firmware |
| 15:40.41 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 15:40.53 | ZX81 | kuyyuk |
| 15:40.55 | ZX81 | yyuku |
| 15:40.57 | ZX81 | yuk |
| 15:40.59 | ZX81 | oops |
| 15:41.11 | ZX81 | :( boohoo |
| 15:41.13 | ZX81 | http://pastebin.ca/24735 |
| 15:41.15 | ZX81 | crashes asterisk |
| 15:41.17 | ZX81 | stupid openh 323 |
| 15:41.19 | ZX81 | :( |
| 15:41.54 | uter | so, if there is anybody who has got blinkenlights on his phone, it would be very helpful to just tell me the firmwareversion |
| 15:42.05 | asterisk99 | malcomd KranZ: Aha!!!! wcte11xp That's the ticket!!!!!!! It seems to work (so far) |
| 15:42.14 | ZX81 | can't anybody puleese help me with http://pastebin.ca/24735 |
| 15:42.27 | KranZ | asterisk99: from now on, you only need to modprobe that |
| 15:42.29 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfll0.dialup.mindspring.com) |
| 15:42.32 | KranZ | it loads zaptel also |
| 15:43.01 | gabb0 | bkw_, wondering if you have a few seconds. just wondering about this squelching of dtmf in meetme. how does this work with PRIs? also what does ZT_FLUSH_ALL do exactly? |
| 15:43.52 | bkw_ | gabb0, you really don't wanna know my honest opinion |
| 15:43.56 | bkw_ | zaptel = poop |
| 15:44.01 | gabb0 | ha |
| 15:44.21 | bkw_ | its fine for small jobs |
| 15:44.25 | bkw_ | but not large jobs |
| 15:45.05 | mogorman | bah |
| 15:45.29 | gabb0 | well I agree, the analog stuff is crap for sure but the t1 cards have been pretty solid for the most part. For us anyway. |
| 15:45.53 | mut | large job = ? |
| 15:45.59 | gabb0 | does the squelching work in meetme on PRIs though? |
| 15:46.36 | *** join/#asterisk file (n=jcolp@mctnnbsa31w-142166094161.nb.aliant.net) |
| 15:47.25 | bkw_ | file |
| 15:47.27 | bkw_ | you there? |
| 15:47.30 | file | yes |
| 15:47.44 | bkw_ | I need you to move nick's sip account to the .12 box in uunet |
| 15:47.48 | bkw_ | and bounce it from there to cogent |
| 15:47.50 | bkw_ | ASAP |
| 15:48.29 | *** join/#asterisk dmg123 (n=mechanix@mechanix.riscom.net) |
| 15:48.41 | file | bkw_: what's the full IP? |
| 15:48.47 | *** part/#asterisk hacim (i=micah@debian/developer/micah) |
| 15:48.49 | bkw_ | look on aim |
| 15:48.56 | file | I'm not on AIM yet lol |
| 15:49.03 | bkw_ | haha |
| 15:49.11 | file | I'm still restoring terminals here |
| 15:49.15 | Katty | bkw_: you never answered me, you know. |
| 15:49.21 | mogorman | see you there blitzrage |
| 15:49.24 | bkw_ | Katty, darth no work here |
| 15:49.29 | Katty | bkw_: i didn't ask that. |
| 15:49.42 | bkw_ | I don't know were darth is ;) |
| 15:49.45 | Katty | k |
| 15:49.56 | tzanger | morning Katty |
| 15:50.03 | Katty | tzanger: hihi (= |
| 15:52.21 | *** part/#asterisk The_Duke (n=The_Duke@80.92.64.103) |
| 15:52.26 | *** join/#asterisk marc324 (n=marc3234@206-248-159-4.dsl.teksavvy.com) |
| 15:52.56 | marc324 | ne1 knows a reliable fax-->email server? |
| 15:55.08 | asterisk99 | malcomd KranZ: Houoston, we have lift off!!!! Missing a D Channel tho |
| 15:55.25 | *** join/#asterisk phidrumdmb (n=email@198.76.96.82) |
| 15:55.29 | phidrumdmb | hey hey |
| 15:55.40 | phidrumdmb | does anyone know how the ac-211 creates call-id numbers? |
| 15:55.55 | phidrumdmb | generates |
| 15:55.57 | phidrumdmb | i mean |
| 15:55.59 | *** join/#asterisk hassler (n=hassler@r-corp.hcst.com) |
| 15:57.34 | hassler | hello folks! I'm not clear on how phone calls are handled on a T1 line vs PRI. I know the PRI is very dynamic, using a dynamic channel for each call (you could have 100 "numbers" coming in on a PRI, as long as no more than 23 simultaneous), but is it the same for the T1, or are the numbers "assigned" to specific channels? |
| 15:58.21 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 15:58.34 | KranZ | t1 is a transport for a pri |
| 15:58.54 | pauldy | t1=pri pri!=t1 |
| 15:59.54 | hassler | I should have said "channelized T1" vs PRI.... Yes, PRI rides on T1 |
| 15:59.55 | fishboy1669 | hi |
| 16:00.12 | fishboy1669 | anyone any idea how to tell if a sip channel has a call active on it |
| 16:00.34 | Ariel_ | show channels |
| 16:00.49 | KranZ | hassler: you mean an analog t1? |
| 16:00.58 | KranZ | to a channelbank |
| 16:00.59 | hassler | no such thing as an analog T1 |
| 16:01.07 | *** join/#asterisk feist (n=feist@nat-pool-msp.redhat.com) |
| 16:01.13 | hassler | yes "channelized T1" could go to a channel bank. |
| 16:01.48 | phidrumdmb | i will revise my question, do some voip telcos use an algo to generate a SIP callid number so phones can not be spoofed |
| 16:04.20 | *** part/#asterisk dmg123 (n=mechanix@mechanix.riscom.net) |
| 16:04.31 | marc324 | does spandsp work ? |
| 16:05.42 | LoRez | supposed to |
| 16:07.33 | *** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se) |
| 16:07.47 | *** join/#asterisk sarfata (n=thomas@did75-11-82-231-43-239.fbx.proxad.net) |
| 16:08.17 | mut | proxad |
| 16:08.19 | mut | ahhhh |
| 16:08.29 | mut | fsckin ppl on that isp annoy me |
| 16:16.14 | *** part/#asterisk trig (n=jb@xob.neospire.net) |
| 16:16.51 | asterisk99 | malcomd KranZ: wewcte11xp loads... but after reboot, and modprobing it again, the LED stays off UNTIL I perform ztcfg -vv |
| 16:17.07 | asterisk99 | malcomd KranZ: wcte11xp loads... but after reboot, and modprobing it again, the LED stays off UNTIL I perform ztcfg -vv |
| 16:18.16 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com) |
| 16:18.37 | *** join/#asterisk christo (n=chris@195.82.114.14) |
| 16:18.59 | christo | does anybody know of a better beep than the standard beep in /var/lib/asterisk/sounds ? :) |
| 16:19.42 | cpatry | christo: just record one with a sexy voice :) |
| 16:20.14 | Beirdo | "better beep"? :) heheh, that's such a subjective thing... |
| 16:20.30 | Beirdo | please leave a message after the scream! |
| 16:20.48 | asterisk99 | malcomd KranZ: How do I get wcte11xp to load automatically on boot? |
| 16:21.07 | Beirdo | OK, lunch time |
| 16:23.27 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:24.55 | christo | hehehe |
| 16:25.01 | christo | these beeps are terrible |
| 16:25.12 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:25.25 | generalhan | whats up guys |
| 16:25.55 | asterisk99 | christo: define terrible ... it's a girlie-man beep? it's too short? it's garbled? |
| 16:26.18 | generalhan | Im having an error here that wont let 2 of my people make calls. everytime they dial it says call failed. And the only message the ocnsole is giving me is "Everyone is congested/busy right now" WTH is going on ? |
| 16:26.29 | ful|work | how can i get CDR variable from agi? i'm using php... |
| 16:26.49 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 16:30.38 | christo | asterisk99 - it's too sharp and not long enough |
| 16:30.48 | christo | I think a beeps should be smooth and long |
| 16:30.51 | christo | *blush* |
| 16:30.55 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:31.34 | *** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
| 16:31.40 | *** part/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
| 16:33.55 | *** join/#asterisk lunk (n=lunk@negative-influence.com) |
| 16:34.57 | asterisk99 | christo: sounds like Bill Clinton's definiton of a beep (Or was that a cigar.... I forget) ;) |
| 16:40.34 | marc324 | ne1 knows of a reliable fax_email server app? |
| 16:41.17 | *** part/#asterisk Attila_Kovacs (n=kovacsat@dsl51B6785D.pool.t-online.hu) |
| 16:41.58 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com) |
| 16:43.45 | *** join/#asterisk Simon- (n=byte@80.193.211.68) |
| 16:44.06 | gabb0 | bkw_, you have worked quite a bit with meetme, or can you tell me who here has. I have a couple more questions about the code |
| 16:45.12 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 16:45.58 | *** join/#asterisk CrazyYoss (n=nobody@adsl-69-236-44-222.dsl.pltn13.pacbell.net) |
| 16:46.14 | devel | anybody have an example static xhtml page that works on the polycom 600 microbrowser? |
| 16:46.17 | ian_k | asterisk99 - get your pri working? |
| 16:48.01 | CrazyYoss | Is there any way to somehow be able to receive MMS messages on the same number you are using for VoIP? Can asterisk figure out what to do with an MMS message and forward it to an application like mbuni? |
| 16:48.22 | JerJer | MMS? |
| 16:48.22 | ian_k | MMS or SMS? |
| 16:48.24 | *** join/#asterisk T0aD (n=toad@epsylon.org) |
| 16:48.29 | CrazyYoss | MMS |
| 16:48.31 | T0aD | <T0aD> good evening everyone |
| 16:48.31 | T0aD | <T0aD> does someone know how to compile openh323 *more* fast ? |
| 16:48.41 | JerJer | don't use Open H.323 |
| 16:48.45 | ian_k | upgrade your system? |
| 16:48.45 | JerJer | that's real fast |
| 16:48.57 | T0aD | oh |
| 16:49.10 | T0aD | this question was not intended to motherfuckers sorry |
| 16:49.31 | devel | ha ha. |
| 16:50.13 | devel | there are very few solutions to make software compile faster. those are pretty valid answers. |
| 16:50.46 | CrazyYoss | I know the difference between SMS and MMS but would that make a difference your response? Can asterisk recognize SMS messages and pass them on? |
| 16:51.10 | JerJer | perhaps nobody knows what MMS is - how about you enlighten us |
| 16:51.16 | ian_k | Toad - do you mean compile the code faster, or compile it so it runs faster? |
| 16:51.32 | *** join/#asterisk Simon- (n=byte@80.193.211.68) |
| 16:51.36 | jarrod | anyone used the presence function with asterisk |
| 16:51.51 | CrazyYoss | SMS has a character limit of 120 characters...MMS allows for more text and attachments such as pictures, sound clips etc |
| 16:51.54 | T0aD | ian_k, i mean to compile it faster |
| 16:52.15 | gabb0 | crazy question, but is there a way to not play dtmf to others in a meetme when the dtmf is inband (other than having that person muted)? |
| 16:52.15 | ian_k | CrazyYoss - yeah, sorta. There is an SMS module for asterisk. If MMS is a real protocol, I don't have any information that it is supported. |
| 16:52.15 | T0aD | cause im still at work and i have to wait for this to finish to compile in order to use my voip stuff.. |
| 16:52.36 | ian_k | toad - how long has it taken? |
| 16:52.55 | T0aD | its compiling the second file |
| 16:52.59 | T0aD | for 20 minutes |
| 16:53.16 | T0aD | oh it just started the third one |
| 16:53.16 | ian_k | toad - something must be broken.. or you have a 286 compiling it.. |
| 16:53.21 | *** join/#asterisk Micc (n=dotirc@c-24-19-175-112.hsd1.wa.comcast.net) |
| 16:53.22 | T0aD | a 500 mhz |
| 16:53.33 | T0aD | nothing is broken, my question is clear |
| 16:53.49 | CrazyYoss | ian_k: thanks ill look into that |
| 16:53.51 | *** join/#asterisk cripito (n=ncripito@67.96.197.99) |
| 16:54.00 | T0aD | ok apparently by default its compiling a lot of crap |
| 16:54.41 | ian_k | toad - hmmm.. does it have -O flags set? |
| 16:55.00 | T0aD | thats a very good question :) |
| 16:55.05 | T0aD | there is a -Os |
| 16:55.21 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
| 16:55.24 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 16:55.36 | T0aD | i guess i can start by removing it |
| 16:55.47 | sahafeez | question about mysql and asterisk if i could.. |
| 16:55.53 | ian_k | it might speed up the compile, but slow down the code that gets compiled |
| 16:56.09 | *** join/#asterisk point (i=1000@213.27.44.55) |
| 16:56.16 | T0aD | yeah i know ian_k |
| 16:56.16 | ian_k | but 20 minutes per file is nuts |
| 16:56.16 | sahafeez | all the docs i have come to read are all about editing the .conf files. |
| 16:56.19 | T0aD | as you said |
| 16:56.27 | sahafeez | i was told i can just put it all, in the db? |
| 16:56.35 | cripito | not all |
| 16:56.36 | cripito | but yes |
| 16:56.50 | sahafeez | the sip.conf and extentions.conf |
| 16:56.52 | cripito | sip - iax cfg, queues, voicemail |
| 16:56.55 | cripito | extensions |
| 16:57.06 | cripito | check the wiki for realtime |
| 16:57.11 | sahafeez | ok. i need to find the docs for that |
| 16:57.14 | sahafeez | realtime. |
| 16:57.14 | sahafeez | ok |
| 16:57.24 | cripito | ;) and u need at least 1.2.X |
| 16:57.27 | T0aD | ian_k, i will try to cross compile it or to compile it distributed but distcc fails to compile such complicated thing |
| 16:57.47 | sahafeez | using HEAD from 2 days ago |
| 16:57.55 | cripito | then it's ok for realtime |
| 16:58.22 | *** join/#asterisk hotgrits (n=hotgrits@192.160.238.156) |
| 16:59.02 | cripito | :) i love this soekris cards sometimes |
| 16:59.36 | *** join/#asterisk justinu (n=j2@72.18.13.40) |
| 16:59.57 | *** part/#asterisk justinu (n=j2@72.18.13.40) |
| 17:00.00 | Primer | anyone here using chan_sccp with a phone behind nat? Seem that skinny (sccp) is quite the crappy protocol. |
| 17:00.09 | *** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp) |
| 17:00.27 | *** join/#asterisk [TK]D-Fender (n=joe@4.67.252.216.dsl1.colba.net) |
| 17:00.53 | *** join/#asterisk justinu (n=j2@72.18.13.40) |
| 17:01.04 | sahafeez | i have a TDM card, FXS and a PRI card. Can I hook my fax to the TDM and map it out the PRI |
| 17:01.36 | *** join/#asterisk Sk3tCh (n=wrickout@86.127.41.114) |
| 17:01.42 | [TK]D-Fender | Got a problem I've been trying to get a handle on : I've got a PRI & Rhino Channel-bank set on * and my faxes often get cut off during receptions. Any tips on how to get them stable? |
| 17:03.15 | jarrod | saha: yes |
| 17:03.23 | sahafeez | cool. |
| 17:03.31 | sahafeez | i will ask how later if i can not figure it out |
| 17:03.39 | jarrod | just have the inbound context on the TDM card dial the extension out the zap driver for the PRI |
| 17:04.14 | cripito | check fax: dial(ZAP/..... also |
| 17:04.29 | jarrod | kinda like exten => _X.,1,Dial(${PRITRUNK}/${EXTEN}) |
| 17:04.30 | cripito | u should be able to redirect when is fax only.. |
| 17:04.54 | sahafeez | and inbound? |
| 17:04.59 | sahafeez | sorry |
| 17:05.01 | sahafeez | got it |
| 17:05.39 | ian_k | there are fax-howto pages on the net. it might be more hassle than its worth though. |
| 17:05.46 | Sk3tCh | how can i make a menu extension?example " push 1 for xxx push 2 for xxx" |
| 17:05.47 | Hmmhesays | woooowooooo tech support calls |
| 17:06.28 | jarrod | sk3tch: record the menu system.. then play it in a context with the exten => 1.. and 2.. defined |
| 17:06.32 | ian_k | Sk3tCh - rtfm @ www.voip-info.org |
| 17:06.45 | marc324 | unrelated-- how to get http file while in linux shell? |
| 17:06.52 | ian_k | wget |
| 17:06.53 | Sk3tCh | it is recorded |
| 17:06.58 | jarrod | marc: fetch / wget |
| 17:07.23 | Sk3tCh | marc324: apt-get install wget , and after wget http://host/file |
| 17:07.25 | sahafeez | ok. just read the real time and dl, maked the add-ons. are there scripts to make the tables in mysql |
| 17:07.25 | ian_k | marc324 - or you can be a man and telnet to remote port 80 directly :) |
| 17:07.38 | jarrod | sk3tch.. NoOp, Background(gsm file), Waitexten.. |
| 17:07.52 | jarrod | then have the exten defined for the menu options that go somewhere |
| 17:07.55 | Sk3tCh | ook |
| 17:08.53 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:09.03 | cripito | they are in the wiki .... if u want i can give u mines but they are a little dif from the originals |
| 17:09.28 | marc324 | gotit |
| 17:09.56 | Sk3tCh | jarrod: and user enters exten and it will redirect to exten, ok...and how can i do this in exten ("press 1 for global language "hu" press 2 for global language "en") |
| 17:10.20 | T0aD | defidev:/var/tmp/openh323_Mimas_patch2# wc -l ./src/h245_3.cxx |
| 17:10.20 | T0aD | 15249 ./src/h245_3.cxx |
| 17:10.22 | T0aD | *erm* |
| 17:10.41 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 17:11.26 | sahafeez | cripito: thanks. reading. it is a bit disorganized |
| 17:11.39 | cripito | true |
| 17:14.10 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:14.32 | *** join/#asterisk myiagy (n=myiagy@200.138.215.78) |
| 17:15.37 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 17:15.49 | Katty | mew. |
| 17:15.50 | harryvv | kinda odd that I went to startup my * and asterisk was not running. Found 10 instances of mpg123 running killed all those and and zaptel driver was no where to be found. Did a recomplile and reinstall and corrected it. |
| 17:16.02 | sahafeez | ok, now i am lost. what order do you do things? is there a check list : setting up asterisk to use mysql 101 |
| 17:16.03 | harryvv | No idea why |
| 17:16.05 | asterisk99 | ian_k: PRI is almost up... my telco guy is saying the D channel is not coming up.... I;m defining signalling=pri_cpe i zappata.conf, but it's not changing anything |
| 17:16.28 | harryvv | hi katty |
| 17:17.06 | *** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
| 17:17.06 | Druken | hi all, has anyone have some experince with a company called netfone ?? |
| 17:17.43 | harryvv | I wonder what kind of power savings I would experaince with a firmware bases asterisk solution for just my own system over that of a old PC that runs it. |
| 17:18.03 | harryvv | Drunken u mean nufone or netfone? |
| 17:18.10 | Druken | netfone |
| 17:18.13 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 17:18.17 | harryvv | never heard of it |
| 17:18.19 | Druken | netfone.ca |
| 17:18.22 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com) |
| 17:18.35 | dca[laptop] | anyone know which flavor of g726 asterisk uses? |
| 17:18.40 | Katty | harryvv: hihi |
| 17:18.49 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 17:18.55 | dca[laptop] | there's G726-32, G726-40, G726-24 on my spa841 available |
| 17:19.08 | harryvv | drunken are you in the greater yvr? |
| 17:19.12 | Ariel_ | g726-32 |
| 17:19.18 | dca[laptop] | Ariel: tks! |
| 17:19.24 | Druken | yvr? |
| 17:19.27 | Ariel_ | harryvv, hello afternoon |
| 17:19.32 | harryvv | yes vancouver |
| 17:19.41 | harryvv | Hi Ariel |
| 17:19.44 | harryvv | :) |
| 17:19.48 | Druken | am i in vancouver? no... |
| 17:19.54 | file | Katty: great idea |
| 17:19.57 | harryvv | okay i am |
| 17:19.58 | harryvv | :) |
| 17:20.01 | Katty | file: let's nap. |
| 17:20.08 | file | Katty: further great idea |
| 17:20.14 | Katty | k |
| 17:20.48 | Druken | nap with nookie, or nap without nookie? |
| 17:20.55 | harryvv | Ariel, moved into my new place. no longer a apt. just to need to work more hours to afford it. |
| 17:21.23 | Katty | Druken: pfft. |
| 17:21.32 | Ariel_ | harryvv, I know that feeling |
| 17:22.08 | Katty | Druken: i shan't go. |
| 17:22.23 | Druken | :P |
| 17:22.38 | Katty | >:( |
| 17:22.48 | macTijn | re |
| 17:22.58 | file | Katty has already shared my internet connection, how much more personal can you get?!? |
| 17:23.24 | *** part/#asterisk point (i=1000@213.27.44.55) |
| 17:24.41 | *** join/#asterisk Micc (n=dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 17:26.05 | tzanger | hahaha |
| 17:26.41 | Sk3tCh | voicemailmain not send the voicemail..why? |
| 17:26.46 | Sk3tCh | i see the log |
| 17:26.56 | *** join/#asterisk malverian[work] (n=pawalls@pawalls.teamgleim.com) |
| 17:27.00 | Sk3tCh | but...nothing about sending ,or sending fail |
| 17:27.21 | malverian[work] | If I use DeadAGI and do a RECORD FILE inside the AGI script.. the AGI should continue running even if the person hangs up? |
| 17:27.44 | marc324 | unrelated-- how to get download realvnc while in shell? |
| 17:27.54 | Sk3tCh | marc324 |
| 17:27.56 | Sk3tCh | wait |
| 17:28.04 | BrianR___ | I gotta rewrite my MWI toggler script :( |
| 17:28.21 | Sk3tCh | rpm or gzip? |
| 17:28.26 | BrianR___ | toggling mwi's on a key system is a pain in the ass.. :( |
| 17:28.35 | marc324 | rpm |
| 17:28.41 | *** join/#asterisk Himeko (n=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 17:29.37 | Himeko | any eyebeam users know offhand where eyebeam is storing account info? |
| 17:29.45 | ian_k | malverian[work] - yes. |
| 17:29.53 | malverian[work] | ian_k, It's not for some reason.. |
| 17:30.15 | *** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1) |
| 17:30.23 | malverian[work] | I'm using: RECORD FILE "filename" "gsm" "#" -1 s=2 |
| 17:30.49 | malverian[work] | If you record and then hang up (without the silence being hit) it gives me a 200 result=0 (hangup) and then doesn't go any further in the AGI script |
| 17:30.49 | ian_k | malverian[work] - hmmm. It is intended to. I don't know much about it, except that it is supposed to continue after hangups. |
| 17:30.53 | *** join/#asterisk Gerriall (n=NonYa@209.42.198.18) |
| 17:30.56 | Druken | are your rollers in the machine white? |
| 17:31.24 | Katty | Lathos42: eep! |
| 17:31.26 | Lathos42 | Druken: No, it gets on the scanner glass and causes a line down the fax |
| 17:31.50 | Druken | well that's a pain in the ass too |
| 17:32.01 | Katty | the drum is a bigger pain |
| 17:32.16 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 17:32.17 | mutilator | anyone ever heard of all tcp/ip traffic dieing from pasting info at a command prompt/telnet session, i have to reboot my pc to get it goin again.. |
| 17:32.32 | file | mutilator: I had a Belkin router that did that, if I pasted something specific |
| 17:32.35 | mutilator | in winxp |
| 17:32.35 | file | it crashed the router |
| 17:32.38 | malverian[work] | But for some reason if it exits because of timeout, it continues. |
| 17:32.50 | mutilator | file: this is weird tho it happens like anything i login to |
| 17:32.56 | file | mutilator: sucky |
| 17:32.59 | *** join/#asterisk morale (n=russell@secure.deadbolt.ca) |
| 17:33.02 | mutilator | i telnet to a cisco router and paste like 4 line commands and my nic dies |
| 17:33.13 | morale | has anyone had luck with asterisk and vonage? |
| 17:33.27 | mutilator | and it's nothing specific either |
| 17:33.31 | mutilator | just any old paste |
| 17:33.35 | harryvv | so do most small voip services use voip wholsale carriers do there local and long distance calls or only long distance. pri her is typically over 1k per month. wonder which is the better of the two. |
| 17:33.36 | file | morale: you can use the softphone account with Asterisk |
| 17:33.57 | malverian[work] | This is odd... |
| 17:34.04 | file | harryvv: small ones usually just resell VoIP traffic because it's cheap |
| 17:34.34 | harryvv | file, even if its local..like routing traffic from canada to the states then back to canada. |
| 17:34.36 | file | if you're just targeting local and have enough people, you can get a PRI with unlimited local in a certain area |
| 17:35.29 | file | harryvv: in that instance it might be cheaper to find a Canadian termination provider... but initially it can certainly be cheaper |
| 17:36.21 | file | all depends how many people you get to start with, and starting amount of income... :) |
| 17:36.40 | InfraRed | whats the kit of choice to connect to E1/Q931 |
| 17:36.41 | harryvv | yea thats the problem |
| 17:36.43 | Cresl1n | psshh |
| 17:36.44 | Cresl1n | income |
| 17:36.49 | Cresl1n | who needs that? |
| 17:37.00 | file | I do!!! |
| 17:37.11 | harryvv | file, the issue is how large a customer base can be aquired just to pay the bills. |
| 17:37.12 | malverian[work] | It's not even saying the script exited... |
| 17:37.17 | Cresl1n | but we work for free right? |
| 17:37.18 | Cresl1n | :-) |
| 17:37.28 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 17:37.35 | Cresl1n | we don't have to pay our bills since we're free software developers |
| 17:37.41 | file | if only... |
| 17:37.42 | Cresl1n | they just disappear for us :-D |
| 17:37.49 | file | mv bills /dev/null |
| 17:37.52 | Sk3tCh | how can i use the pressed (dtmf)number in tel? |
| 17:38.01 | Sk3tCh | some cmd for it? |
| 17:38.11 | Cresl1n | sk3tch: huh? |
| 17:38.20 | file | I would assume you press the key on the telephone |
| 17:38.26 | file | call me crazy, but I think that's what you do |
| 17:38.34 | wunderkin | file: you're crazy |
| 17:38.38 | file | I know that! |
| 17:38.43 | wunderkin | are you high? |
| 17:38.47 | file | no :( |
| 17:38.49 | harryvv | Ohh man I have to laugh at this site. For some time the northtel.com web site was for sale. Type in northtel.com and see what domain it directs to :) |
| 17:38.50 | file | I'm file! |
| 17:39.28 | wunderkin | heh |
| 17:39.32 | Sk3tCh | Cresl1n: i called a number ok..and i press btw button 4 then it speak something.. how can i get the typed number? |
| 17:39.53 | Cresl1n | in your dialplan |
| 17:40.10 | harryvv | I said to the owner "because this site is very close to the name nortel network aka northern telecom that site is a trade name violation and thus, I dont want to be sued" |
| 17:40.11 | harryvv | :) |
| 17:40.29 | Sk3tCh | Cresl1n: ok but what is the command to get? |
| 17:40.36 | Sk3tCh | getdtmf or something? |
| 17:40.51 | Cresl1n | just make an extension that has that number |
| 17:40.52 | harryvv | wunderkin so you see, thay listened to my advice :) |
| 17:41.15 | file | Background will play a file and listen for digits, it then sees if there's an extension in the current context that matches the digits |
| 17:41.16 | file | mmmkthx |
| 17:41.21 | harryvv | you mean nortelnetwoks.com |
| 17:41.22 | harryvv | :) |
| 17:41.23 | file | Lathos42: a networked wok, duh |
| 17:41.51 | Lathos42 | harryvv: Well, I figured that a Nortel Netwok is just a netwok sold by Nortel.. :) |
| 17:41.53 | *** join/#asterisk malabar (n=mala@bkkb-gw.bitcon.no) |
| 17:42.12 | Lathos42 | file: Does it make it so you can check your email while making dinner? |
| 17:42.23 | file | yup |
| 17:42.49 | harryvv | file, you have a service right? |
| 17:43.01 | file | do I? personally? uh no |
| 17:43.09 | file | rephrase your question. |
| 17:43.25 | harryvv | voip service |
| 17:43.26 | harryvv | :) |
| 17:43.31 | shido6 | nortel ewoks |
| 17:43.40 | file | I work for Asterlink which provides toll-free origination and termination services |
| 17:43.44 | *** join/#asterisk andrewsbenjamin (n=chatzill@miro.voltaiccommerce.com) |
| 17:43.49 | harryvv | right |
| 17:44.08 | Lathos42 | file: You corporate shill |
| 17:44.11 | harryvv | file, that is a per min and or monthly rate? |
| 17:44.12 | harryvv | :) |
| 17:44.14 | file | per min |
| 17:44.17 | harryvv | XO, Stealth hook up in gated community of VoIP carriers seeking to bypass the public network and create a parallel Internet. |
| 17:44.33 | harryvv | mmm off news from voip-info.org |
| 17:44.38 | harryvv | odd news |
| 17:45.38 | marc324 | where can i find a fax->email server apps? |
| 17:45.49 | InfraRed | marc324: heard of google.com ? |
| 17:46.04 | InfraRed | ~docs |
| 17:46.06 | jbot | i heard 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 |
| 17:46.10 | InfraRed | marc324: ^^^^^ |
| 17:47.20 | *** part/#asterisk dca[laptop] (n=dca[lapt@sta-206-168-218-202.rockynet.com) |
| 17:47.47 | *** join/#asterisk Mike (n=mike@201.135.48.190) |
| 17:47.47 | *** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net) |
| 17:48.08 | Micc | I've tried firefly and iaxcomm now and both have problems skipping audio. |
| 17:48.09 | Mike | guys its there a howto on how to conect to a carrier using g729 without buying a license on my side? |
| 17:48.41 | harryvv | sounds like jitter |
| 17:49.07 | Micc | harryvv, how can I fix the jitter? It seems to have the same problem with IAX or SIP. |
| 17:49.30 | Beirdo | do what I tell you, you bitch! |
| 17:49.33 | FuriousGeorge | our bookkeeper is neurotic (im not kidding) |
| 17:49.35 | Micc | harryvv, my round trip time to the server is always less than 40ms. |
| 17:49.54 | FuriousGeorge | i just put accounting on an * system, and one of the crap handsets on the zap channels has echo |
| 17:50.12 | FuriousGeorge | she insists on using that phone |
| 17:50.45 | FuriousGeorge | she starts to explain why and it just goes into this heavily accesnted english jibber jabber, like someone on LSD talking to you |
| 17:50.58 | Micc | Even when I run directly to nufone it has the same problem. |
| 17:51.03 | FuriousGeorge | the point is |
| 17:51.47 | FuriousGeorge | i think i gotta lower or raise one if the gains on that channel because i hear echo when people raise their voices on the outseide asterisk end |
| 17:52.20 | FuriousGeorge | to be more exact, outside party hears themselves when they shout, which gain do i gotta lower raise? rx? |
| 17:52.31 | FuriousGeorge | would -1 make it quiter? |
| 17:52.37 | FuriousGeorge | quieter? |
| 17:52.39 | *** part/#asterisk Micc (n=dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
| 17:53.00 | FuriousGeorge | (new handsets are in the mail) |
| 17:53.10 | FuriousGeorge | (will be) |
| 17:54.10 | harryvv | rxgain=0.0 |
| 17:54.10 | harryvv | txgain=0.0 |
| 17:54.14 | FuriousGeorge | i cant see if you raise your hands just shout it out when you know it |
| 17:54.15 | harryvv | thats my settings. |
| 17:54.36 | FuriousGeorge | harryvv: that works great with echocancel and echocancel when bridged |
| 17:54.36 | harryvv | FuriousGeorge is this on a zap card? |
| 17:54.38 | FuriousGeorge | ya |
| 17:54.59 | FuriousGeorge | but this one particular handset has an echo |
| 17:55.20 | harryvv | I hear there is some very good ballanced bridge gateways on the market that will impedence match the lines thus ridding of the echo. |
| 17:55.56 | FuriousGeorge | so if surmising if i lower the rxfain to -1.0 or something the calling party's recieved voice will be made a bit quieter so he would need to shout louder |
| 17:56.05 | FuriousGeorge | harryvv: my echo cancel works great on everything else |
| 17:56.31 | FuriousGeorge | i dont wanna buy more hw if i dont have to, and im ordering new handsets today, which brings me to another question: |
| 17:56.51 | FuriousGeorge | are there any analog phones which have any features that are particularly * friendly |
| 17:56.59 | FuriousGeorge | like programamble park and xfer buttons |
| 17:57.32 | harryvv | fur, well we use the uniden wireless hooked to the sipura ata and it works fine. |
| 17:57.41 | harryvv | put on hold transfer ect. |
| 17:58.15 | FuriousGeorge | harryvv: ill be looking at those on froogle |
| 17:58.29 | harryvv | froogle ? |
| 17:58.36 | marc324 | is everyone running asterisk using shell or in xwin? |
| 17:58.41 | FuriousGeorge | harryvv: www.froogle,com its great if ur in the us |
| 17:58.56 | Qwell | yeah, froogle is awesome |
| 17:59.02 | marc324 | for configuring |
| 17:59.05 | Qwell | except, I almost always end up buying stuff at newegg, heh |
| 17:59.07 | FuriousGeorge | marc324: i connect to the console in konsole if thats what you mean |
| 17:59.18 | FuriousGeorge | Qwell: me too |
| 17:59.28 | FuriousGeorge | Qwell: but there are some things even newegg doesnt have |
| 17:59.34 | FuriousGeorge | like tdm and phones |
| 17:59.35 | Qwell | FuriousGeorge: yep, I've found a few things |
| 17:59.55 | Qwell | specific "video card" from asus, for an intel 915 chipset |
| 17:59.57 | harryvv | ohh really froogle looks up a item by make and model and list the item that cost the least? sounds like pricewatch |
| 18:00.15 | FuriousGeorge | harryvv: more inclusive than pricewatch |
| 18:00.16 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 18:00.18 | Qwell | harryvv: but pricewatch isn't that great, and there is one major difference |
| 18:00.23 | Qwell | people submit to pricewatch |
| 18:00.25 | Qwell | froogle crawls |
| 18:00.28 | FuriousGeorge | harryvv: i switched like 2 years ago |
| 18:00.39 | FuriousGeorge | and what he said |
| 18:00.52 | Qwell | I'll admit though, pricewatch was very good at one point |
| 18:00.55 | harryvv | FuriousGeorge switched to froogle from pricewatch? |
| 18:01.04 | harryvv | k |
| 18:01.04 | Qwell | and really, they still are good...there is just somebody better |
| 18:01.25 | harryvv | I love newegg. most of my new hardware is from them. |
| 18:01.27 | FuriousGeorge | harryvv: but sometimes when i just want mb or cpu or especially mb+cpu quick quotes i shoot to pricewatch |
| 18:01.36 | harryvv | sure |
| 18:01.48 | Qwell | FuriousGeorge: Thats one benefit of pricewatch - since people submit to it, it has stuff like mb+cpu combos |
| 18:02.22 | FuriousGeorge | Qwell: do you ever have to build legacy boxes? if so what do you use. |
| 18:02.22 | Ariel_ | marc324, I run asterisk as a server without xwindows |
| 18:02.30 | Qwell | legacy? no, heh |
| 18:03.22 | FuriousGeorge | sometimes i build all or part of a linux nat box from parts |
| 18:03.36 | Qwell | Lathos42: for a much higher price |
| 18:03.58 | harryvv | I dont know if I will buy used or never used hardwar from ebay again. One guy advertised a amd opteron mobo and was working. Well it would lock up. After futsing for a long time to get it working it was the mobo its self not the other hardware. the guy could not be contacted and yet he had a good rating. its going to be new next time for me. |
| 18:04.05 | *** join/#asterisk fiber0pti (n=johndoe@207.114.199.98) |
| 18:04.06 | Lathos42 | file: I hope you charge the actual cost to ship me, i'm a big guy :) |
| 18:04.21 | *** join/#asterisk xon-xoff (n=optikal@64.80.3.62) |
| 18:04.26 | file | freight! |
| 18:04.46 | Lathos42 | Qwell: My dell rep came in lower than everyone else on all of the parts that i've specced for our phone system |
| 18:04.59 | FuriousGeorge | and there is no point in using even a $20 64mb agp 4 gpu when u can get a $10 pci video card. sometimes i use isellsuprolus.com but the prices arent /that/great |
| 18:05.09 | FuriousGeorge | i just hate buying important parts 2nd hand |
| 18:05.13 | FuriousGeorge | from ebay |
| 18:05.25 | Qwell | FuriousGeorge: check out...umm...shit, one sec |
| 18:05.29 | FuriousGeorge | *isellsurpluss.com |
| 18:05.36 | Qwell | are you in the US? If so, is it CA? |
| 18:05.42 | FuriousGeorge | other side |
| 18:05.44 | FuriousGeorge | ~nj |
| 18:05.45 | jbot | from memory, nj is home to the Sopranos |
| 18:05.51 | Qwell | k, check Redemtech |
| 18:05.56 | Qwell | lemme see if I can find a link |
| 18:06.07 | Qwell | (it's a pita to find their store) |
| 18:06.13 | *** join/#asterisk Tili (i=Tili@202-133-67-210-dialup.sat.net.pk) |
| 18:06.38 | FuriousGeorge | i found one |
| 18:07.03 | FuriousGeorge | that advertises as "help you recycle and refurbish your old hw" consultants |
| 18:07.13 | FuriousGeorge | " at least it appears at a glance |
| 18:07.17 | Qwell | FuriousGeorge: http://www.redemtech.com/webstore/Pages/PersonalizeHomePage.aspx |
| 18:07.25 | Qwell | They have decent prices on "recycled" machines |
| 18:09.00 | Qwell | FuriousGeorge: this place basically gets a call from a company, "Hey, I've got some old* machines here, can you come pick them up?" |
| 18:09.15 | Qwell | *not always that old |
| 18:09.23 | Qwell | Then they just turn around and resell them |
| 18:10.10 | FuriousGeorge | Qwell: this will come in usefull |
| 18:11.00 | marc324 | is vnc ok for accessing asterisk from xp? |
| 18:11.06 | Qwell | no! |
| 18:11.08 | Qwell | use ssh |
| 18:11.13 | Qwell | X on a server is plain dumb |
| 18:11.26 | Ariel_ | or putty |
| 18:11.28 | marc324 | why? |
| 18:11.50 | FuriousGeorge | clear text passwords everywhere |
| 18:11.52 | marc324 | are you saying not to install x? |
| 18:11.58 | Qwell | yes |
| 18:12.11 | Ariel_ | No xwindows |
| 18:12.19 | FuriousGeorge | marc324: why you need x? |
| 18:12.33 | marc324 | for editing the config files |
| 18:12.37 | harryvv | marc, use putty |
| 18:12.57 | Ariel_ | harryvv, I said that... But also ssh has a nice setup for windows... |
| 18:13.01 | FuriousGeorge | i know on gentoo, emerging asterisk can pull X in unless you specify -X in make.conf, for some feature i dont know about |
| 18:13.23 | FuriousGeorge | marc324: i like nano, thats like the non-tchy vi |
| 18:13.28 | JerJer | FuriousGeorge: that's a fuckup on the part of whoever created the gentoo packaging |
| 18:13.31 | FuriousGeorge | non-techy* |
| 18:13.38 | harryvv | Ariel_ so your saying ssh can be used on the command line in xp? |
| 18:13.51 | FuriousGeorge | JerJer: fools! |
| 18:14.03 | Ariel_ | harryvv, no there is a co. that makes ssh software for windows to connect to a linux box. |
| 18:14.11 | FuriousGeorge | harryvv: sure, from the cygwin commandline |
| 18:14.14 | *** join/#asterisk mohr_ (n=Christia@host8.itech.is.ew.ro) |
| 18:14.16 | SwK[Work] | harryvv: putty is a windows ssh client |
| 18:14.31 | FuriousGeorge | or you can putty (thats what i do) |
| 18:14.37 | harryvv | Swk, yes and i putty into my asterisk all the time. |
| 18:14.37 | SwK[Work] | google for putty win32 and hit 'i'm feeling lucky" |
| 18:14.53 | FuriousGeorge | harryvv: google (feeling lucky) putty download |
| 18:14.58 | FuriousGeorge | lol |
| 18:15.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:15.09 | *** part/#asterisk zoo (i=nobody@ip-98-16.travedsl.de) |
| 18:15.29 | Ariel_ | Putty good |
| 18:15.35 | cripito | i use ssh client from www.ssh.com |
| 18:15.45 | FuriousGeorge | harryvv: cygwin installs some linux api or something on your windows box, if your using to use the openssh client itself |
| 18:15.47 | cripito | is way better than putty but putty is ok |
| 18:15.50 | Ariel_ | cripito, yes it's a good product and has built in SCP3 |
| 18:15.55 | FuriousGeorge | ud need to install a few packages |
| 18:16.11 | cripito | we agree ariel |
| 18:16.19 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 18:17.37 | PupenoL | When RTP is mentioned in the context of Asterisk, does it refeer to http://www.ietf.org/rfc/rfc3550.txt ? |
| 18:17.47 | sahafeez | is there a GUI that supports Realtime w/mysql for adding + configuring users |
| 18:18.20 | marc324 | whats puttytel? |
| 18:18.28 | FuriousGeorge | these uniden handsets are too expensive, plus wont they have a bunch of features that overlap *? im looking for a simple 1 line corded handset, with a programamble button or two |
| 18:18.30 | cripito | sahafeez i am building 1.... |
| 18:19.12 | FuriousGeorge | like if i could program a park button, that would be cool |
| 18:20.00 | *** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net) |
| 18:20.49 | *** join/#asterisk mutilator (n=animenod@65.111.201.79) |
| 18:20.53 | sahafeez | cripito: so no... |
| 18:20.56 | Tili | what is the tool name to capture rtp traffic on a network and save it in a file |
| 18:21.37 | cripito | check this out http://www.cripiland.com/screenshots/manager1.jpg |
| 18:21.56 | cripito | and this http://www.cripiland.com/screenshots/manager2.jpg |
| 18:22.08 | sahafeez | ok. can i dl and use it? |
| 18:22.11 | cripito | so the answer is a complete realtime product no |
| 18:22.18 | cripito | no yet.. |
| 18:22.29 | cripito | soon |
| 18:23.20 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 18:23.22 | cripito | i am almost there... |
| 18:23.37 | cripito | but there is web appl that do this.. check the wiki |
| 18:23.38 | *** join/#asterisk gandhijee (n=gandhije@host-66-202-34-162.spr.choiceone.net) |
| 18:23.56 | gandhijee | can someone give me a hand plz? |
| 18:24.07 | gandhijee | i just got a TDM400 w/ 1 FXO and 3 FXS |
| 18:24.14 | gandhijee | thanks george |
| 18:24.17 | FuriousGeorge | always wanted to do that |
| 18:24.31 | FuriousGeorge | no, thank you :) |
| 18:24.32 | cripito | give a hand? |
| 18:24.38 | cripito | :D |
| 18:24.38 | gandhijee | and when i modprobe wcfxo is kinda say "up yours buddy" |
| 18:24.44 | MattH | Hi. I'm trying to figure an issue out here... if a sip phone has 'forwarding' on, when the asterisk box goes to call the sip phone the sip phone says 'HEY! I MOVED!' and directs the call elsewhere. Well now the call goes out ZAP/SIP/IAX trunk but it doesn't log as going to or from that user.... is there some way I can correct this? The account code field just gets totally blanked (assumably because the call is not coming FROM the sip phone), and the |
| 18:24.44 | MattH | <PROTECTED> |
| 18:25.00 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
| 18:25.14 | FuriousGeorge | gandhijee: it doesnt say that |
| 18:25.23 | MuppetMaster | Hello everyone. Any one know how to get Asterisk to connect to an Avaya CM station side per this post: http://forums.digium.com/viewtopic.php?p=5307#5307? |
| 18:25.36 | gandhijee | http://pastebin.com/385270 |
| 18:26.01 | gandhijee | FuriousGeorge: true its says what it does on this pastebin |
| 18:26.25 | gandhijee | but i can load the wcfxs with no problem... and it activates the 4 ports... |
| 18:26.43 | cripito | gandhijee... can u paste also ur dmesg (the last part) |
| 18:27.20 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
| 18:27.33 | FuriousGeorge | gandhijee: im no expert, but i only modprobe wxfxs and i use 2fxo and 2 fxs |
| 18:27.42 | gandhijee | oh shit |
| 18:27.46 | FuriousGeorge | and it works, so maybe you dont need it |
| 18:27.59 | gandhijee | i'm a fucking retard, apparently the wcfxs handles the FXO too?? |
| 18:28.10 | Qwell | gandhijee: use wctdm |
| 18:28.27 | Qwell | gandhijee: wcfxs was renamed to avoid that confusion, iirc |
| 18:28.27 | harryvv | gand, what errors you getting? |
| 18:28.35 | gandhijee | qwell: don't seem to have that one |
| 18:28.43 | Qwell | might be a cvs thing... |
| 18:28.48 | gandhijee | qwell: im using zaptel-1.0.9.2 |
| 18:29.19 | Qwell | but yeah, wcfxs == wctdm |
| 18:29.19 | gandhijee | but FuriousGeorge was right, just modprobing the wcfxs wored |
| 18:29.28 | gandhijee | *worked |
| 18:29.34 | wunderkin | Cresl1n are you available? |
| 18:30.07 | gandhijee | yea it work |
| 18:30.07 | gandhijee | http://pastebin.com/385278 |
| 18:30.10 | gandhijee | i didn't think about checkin dmesg to see what it had to say to me |
| 18:30.19 | gandhijee | i was just remembering from the old how-to's |
| 18:31.07 | *** join/#asterisk bawks (n=0x3e44d@67-41-182-243.slkc.qwest.net) |
| 18:31.49 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 18:32.07 | bawks | Is it possible to set up TE110P into passive mode where it could just listen to the traffic going across the PRI via a splitter? |
| 18:32.20 | Qwell | bawks: for what purpose? |
| 18:32.32 | bawks | to CDR a PBX that doesnt have CDR |
| 18:32.49 | Cresl1n | wunderkin: what's up? |
| 18:33.12 | *** join/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com) |
| 18:34.17 | wunderkin | Cresl1n: hey dude, i just have a quick question, i'm getting the finger pointing.. i have 2 pri from the same carrier but diff resellers, on the initial install (on a single t1 card/diff box) they worked.. 1st one installed ok then installed 2nd and they said it was configured wrong there.. now its ok.. so put in 4 port card diff box, the 1st one wont ever come up.. it says Provisioned, In Alarm, Down, Active, they are seeing remote made |
| 18:34.27 | wunderkin | unfortunately i never tested the 1st one again after the 2nd was installed |
| 18:34.42 | wunderkin | theyve done all their testing that they can do and they say its all clean but im remote made busy |
| 18:35.08 | wunderkin | is it possible for them to have the same numbers as the other and so one works and the other shows busy? niu on both is green and my side is red alarm |
| 18:35.34 | justinu | i hate that nortel remote made busy shit |
| 18:35.35 | wunderkin | now he sais carrier fail |
| 18:35.50 | MattH | i hate nortel |
| 18:35.51 | justinu | can you loop it up for him at the card? |
| 18:36.33 | FuriousGeorge | can anyone recomend a basic analog wired handset to use with *. the feature that i'm looking for is a programable park button |
| 18:37.56 | justinu | also, have you tried putting a loop up facing your PRI card, to see if you can see your own carrier? |
| 18:38.09 | Cresl1n | wunderkin Are their red alarms on the line? |
| 18:38.38 | hardwire | wunderbumpkin |
| 18:38.39 | wunderkin | its red for me green on niu |
| 18:38.57 | Cresl1n | it should be green for you |
| 18:39.02 | Cresl1n | if it's red, then you have a problem |
| 18:39.07 | Cresl1n | layer one is down |
| 18:39.17 | Cresl1n | are you sure that you configured it right in zaptel.conf? |
| 18:40.07 | wunderkin | the working circuit works on either port, same config |
| 18:40.38 | justinu | perhaps they misoptioned the circuit at the mux/smartjack |
| 18:40.40 | wunderkin | sorry if i dont answer all questions im on the phone with tthem now |
| 18:40.51 | justinu | try coming up to D4 framing |
| 18:41.01 | wunderkin | the lec tested the smartjack to the patch in my rack and works ok |
| 18:41.17 | Cresl1n | wunderkin: is the working circuit in alarm? |
| 18:41.35 | T0aD | is it ok to use gnugk 2.2.1 with openh323 1.15.3 ? |
| 18:41.45 | Cresl1n | wunderkin: look under zttool with the lines plugged in and see what the status is for each line |
| 18:41.56 | wunderkin | at one point they said it was up but some problem w/ me |
| 18:42.00 | T0aD | do you have good results on it ? i have a strange issue while redirecting numbers to gk h323 endpoints |
| 18:42.05 | justinu | they're always going to blame you |
| 18:42.09 | wunderkin | i know |
| 18:42.25 | justinu | you need to use loops to find where the problem is |
| 18:42.33 | justinu | do you have loopback plugs? |
| 18:42.36 | wunderkin | yeah |
| 18:43.02 | justinu | so plug the rj45 that plugs into your zap card into a loopback plug |
| 18:43.03 | dabigshiznizzle | take the cable from your port and loop tx and rx back to yourself. |
| 18:43.07 | wunderkin | red alarm |
| 18:43.12 | justinu | that'll put up a loop towards your carrier |
| 18:43.16 | justinu | does the circuit come up? |
| 18:43.28 | wunderkin | before when i put ha rd loop before the equip and it was ok |
| 18:44.18 | *** join/#asterisk websae (n=root@206.132.218.42) |
| 18:44.19 | justinu | have them do BER testing on the circuit when the loop towards them is up at the very end of the circuit |
| 18:44.36 | wunderkin | yesterday they did a whole lot of testing |
| 18:44.46 | justinu | that'll test the entire length of the run, perhaps there is a problem in your DSX patch or cable |
| 18:44.51 | wunderkin | and the lec tested from the niu to me when there was a loop on that side i guess |
| 18:45.06 | websae | I have a question...I am running an asterisk server...my sip phone (sipura 841 hard phone) registers fine, can make a call, person can hear me, but I can't hear them...anyone have any ideas? |
| 18:45.12 | wunderkin | and the other thing is that they switched the cross connects since i have 2 and the one that works is ok on either one |
| 18:45.21 | bawks | Is there a way to log all console message to a file? |
| 18:45.26 | bawks | messages |
| 18:46.01 | justinu | i dunno how the lec could test from the NIU to your equip |
| 18:46.07 | justinu | if you didn't put up a loop for them |
| 18:46.08 | wunderkin | tbird |
| 18:46.13 | websae | if anyone could help that'd be great!! :) |
| 18:46.15 | wunderkin | there was a serviceman there this morning |
| 18:46.15 | *** part/#asterisk dabigshiznizzle (n=dabigshi@gwsecure.sctelcom.com) |
| 18:46.23 | websae | i don't know why i can make a call and they can hear me, but i can't hear them |
| 18:46.33 | justinu | when he ran ESF to your zaptel, did the carrier come up for you? |
| 18:46.43 | wunderkin | i wasnt there during the service |
| 18:47.00 | wunderkin | i know when they do their testing i do see it says it comes up but when their testing is done it goes back down |
| 18:47.03 | justinu | then go to the other end of your cable, that plugs into the DSX panel, and put the loop up facing you |
| 18:47.11 | wunderkin | and i get a lot of hdlc aborts when they do that but im not sure what k ind of test they did |
| 18:48.24 | justinu | ask your carrier to put a loop up at their switch facing you as well |
| 18:48.43 | justinu | if you can sync up to your own carrier, the problem is their switch is misoptioned. |
| 18:49.15 | websae | Anyone at all...have any ideas why when i call out with my phone, i can connect the call, they can hear me, but i can't hear them... |
| 18:49.19 | websae | anyone ;)?? |
| 18:49.30 | justinu | web, it sounds like a NAT issue |
| 18:49.49 | justinu | are you connecting to asterisk on a private NATed lan? |
| 18:49.53 | websae | yes |
| 18:50.03 | justinu | are both the phone and the PBX behind the firewall? |
| 18:50.03 | websae | but connecting to my friend's asterisk server it had worked |
| 18:50.22 | websae | the PBX is at a remote location with it's own static ip address |
| 18:50.33 | justinu | is STUN turned on? |
| 18:50.35 | justinu | on the phone |
| 18:50.41 | mutilator | man i have the weirdest issue going on right now |
| 18:50.42 | websae | the phone is at my house, behind a linksys router |
| 18:50.53 | websae | it's a SIPURA 841---i don't think it has STUN |
| 18:50.56 | justinu | it does |
| 18:51.15 | websae | i don't know how i would turn that on or off |
| 18:51.21 | websae | i have never messed with that setting on it |
| 18:51.26 | justinu | in the web interface |
| 18:51.35 | justinu | try turning nat mapping enable to on |
| 18:51.37 | hardwire | who loves me here? |
| 18:51.39 | hardwire | really |
| 18:51.40 | justinu | and nat keep alive enable to on |
| 18:51.42 | websae | and before my phone worked fine on my friend's asterisk server which was behind a firewall, just had DMZ |
| 18:51.43 | hardwire | who wants to give me some love |
| 18:51.47 | mutilator | i do! |
| 18:51.50 | mutilator | anyone ever used a cisco as5350? |
| 18:51.55 | websae | turn nat mapping enable = on? |
| 18:51.58 | hardwire | mutilator: I have seen it |
| 18:52.01 | hardwire | wanted to shoot one |
| 18:52.01 | justinu | yeah |
| 18:52.13 | mutilator | well |
| 18:52.26 | hardwire | mutilator: contact Arsenal on slashnet |
| 18:52.31 | hardwire | he are smart. |
| 18:52.44 | hardwire | hmm.. he are also offline |
| 18:52.48 | websae | how about NAT KEEP ALIVE? |
| 18:52.53 | websae | should that be on as well? |
| 18:52.55 | mutilator | problem i'm having... when a call comes in on the pri, if the callerid is 6851070 then the call fast busies, doesn't even show up on my asterisk debug |
| 18:53.04 | mutilator | but if the callerid is 6851079 or something else |
| 18:53.06 | mutilator | the call works fine |
| 18:53.22 | mutilator | i run a debug on the cisco box and as far as i can tell the call is being sent to asterisk |
| 18:53.25 | websae | in the asterisk server....should just have NAT=1 & QUALIFY=NO ? |
| 18:53.26 | justinu | websae: probably |
| 18:53.29 | mutilator | but the log is kinda cryptic |
| 18:53.42 | BrianR___ | Hmm.. I think I've discovered a case where SetGlobalVar doesn't work :( |
| 18:53.42 | hardwire | callerid? |
| 18:53.43 | hardwire | or |
| 18:53.45 | hardwire | did? |
| 18:53.49 | mutilator | did |
| 18:53.50 | justinu | in asterisk sip.conf, nat=yes, qualify=no |
| 18:53.56 | hardwire | mutilator: well |
| 18:53.59 | morale | whats a good voip provider to go with in canada.. or more like alberta that supports SIP? |
| 18:54.03 | justinu | mutilator: how do you know the call is actually hitting your PRI? |
| 18:54.06 | hardwire | maybe you should set up that extension in asterisk to set to Echo() |
| 18:54.09 | froud | Google has no fear: From: http://www.theinquirer.net/?article=26734 |
| 18:54.11 | mutilator | as i said |
| 18:54.21 | mutilator | far as i can tell is in the cisco the call is being sent to asterisk |
| 18:54.28 | mutilator | so that'de mean it's hittin the pri |
| 18:54.43 | websae | what is NAT=1 then? |
| 18:54.46 | justinu | no idea |
| 18:54.50 | websae | isn't that the same NAT=yes? |
| 18:54.52 | ender | mutilator: only the callerID changes, not the 'calledid' ? |
| 18:54.57 | mutilator | yea |
| 18:55.03 | mutilator | even if the calledid changes |
| 18:55.12 | wunderkin | loop on niu and get hdlc aborts, carrier up, unable to loop csu |
| 18:55.16 | ender | thats f'd up. Sounds like a telco issue. |
| 18:55.21 | mutilator | anything sent to the asterisk box is fast busied long as my callerid is 6851070 |
| 18:55.33 | KranZ | anyone had inbound callerid issues with a dvg-1402s? |
| 18:55.36 | mutilator | not a telco issue cause it's an internal call |
| 18:55.42 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 18:55.48 | ender | mutilator: hrm. |
| 18:56.04 | ender | mutilator: so all you're doing is overriding the $CALLERID vairable? |
| 18:56.10 | mutilator | nope |
| 18:56.16 | mutilator | asterisk is the end point in all this |
| 18:56.45 | mutilator | doesn't even seem to get there, my cisco dump says something to the effect it's sent but i can't really tell what it's doing |
| 18:56.52 | ender | mutilator: yes, I know, but you can in your dialplan override a callerID variable. |
| 18:57.30 | mutilator | in my pbx i change the callerid of the phone on my desk |
| 18:57.40 | mutilator | then i call a voip number which goes through the cisco to asterisk |
| 18:57.45 | ender | what is the path here? |
| 18:57.58 | mutilator | if the callerid of the phone on my desk is 6851070 then i fast busy |
| 18:58.01 | mutilator | if not then it works fine |
| 18:58.04 | mutilator | goes all the way to my voip |
| 18:58.17 | ender | deskphone -> ? |
| 18:58.34 | mutilator | deskphone -> pbx -> adtran -> cisco -> asterisk -> ata |
| 18:58.50 | KranZ | cisco router? |
| 18:58.52 | ender | wuf |
| 18:58.56 | mutilator | yes as5350 |
| 18:59.12 | KranZ | is the adtran running voip? |
| 18:59.15 | ender | mutilator: do you hav emultiple ATA phones? can you use one ATA phone to call the ext of another ATA phone? |
| 18:59.23 | mutilator | ata to ata works fine |
| 18:59.35 | mutilator | even with callerid 6851070 on one of em |
| 18:59.42 | ender | mutilator: ah, that was my next ? |
| 18:59.53 | ender | so it still seems that * isn't balking on the caller ID |
| 19:00.25 | KranZ | maybe the cisco is blocking b/c it doesnt allow spoofed callerids? |
| 19:00.28 | mutilator | i put a exten => _. in my dialplan to noop callerid to see if it even entered |
| 19:00.44 | Ariel_ | yeppiee more traffic in 2010. Miami is to host the 2010 SuperBowl |
| 19:00.45 | mutilator | doesn't even show anything |
| 19:00.48 | KranZ | mutilator: the s extension does the same |
| 19:00.50 | mutilator | KranZ: if i cahnge my callerid to something else it works fine |
| 19:00.59 | mutilator | ya |
| 19:01.06 | wunderkin | is it a problem if my pri carrier loops the niu and i get hdlc aborts? they see carrier up but they are unable to loop the csu |
| 19:01.06 | Qwell | Ariel_: just hope it isn't your team |
| 19:01.19 | KranZ | mutilator: you're verbosity is up right |
| 19:01.19 | Ariel_ | Qwell, I hope to be out of here then |
| 19:01.27 | KranZ | you see an attempt to place the call? |
| 19:01.37 | Qwell | Ariel_: with basketball, we get riots |
| 19:01.45 | Qwell | win or lose, actually |
| 19:02.24 | hardwire | hmm |
| 19:02.26 | hardwire | smokeping is nice |
| 19:02.32 | mutilator | ya |
| 19:02.33 | mutilator | at 5 |
| 19:02.37 | hardwire | think I will use the udp echo tests |
| 19:02.39 | hardwire | vs icmp |
| 19:02.48 | websae | justin: you still here? |
| 19:02.54 | mutilator | hmm |
| 19:03.01 | mutilator | lemme try this |
| 19:03.10 | mutilator | call w/ caller id 1070 and 1079 |
| 19:03.17 | mutilator | and diff both ccapi debugs of the cisco |
| 19:03.19 | mutilator | see what changes |
| 19:03.47 | websae | under NAT SUPPORT PARAMETERS in my sipuara 841 config, talks about Handle VIA recieved...Insert VIA receieved...Handle VIA rpport....Insert VIA rport...anyone have any ideas? |
| 19:03.53 | websae | what those VIA settings are? |
| 19:04.00 | websae | under NAT Support Parameters? |
| 19:05.24 | hardwire | generating thumbnails for 700 + images sucks |
| 19:05.28 | hardwire | in nautilus |
| 19:05.42 | harryvv | nautilus? |
| 19:05.51 | hardwire | naughtyless |
| 19:05.55 | harryvv | :) |
| 19:06.13 | *** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) |
| 19:06.15 | Ariel_ | Qwell, your in Detroit |
| 19:06.18 | hardwire | ok |
| 19:06.19 | Qwell | Ariel_: LA |
| 19:06.24 | hardwire | I am going to attempt replacing the processor in my laptop |
| 19:06.37 | Ariel_ | Qwell, ahh |
| 19:06.43 | hardwire | ender: I am pick and choosing images |
| 19:06.44 | Qwell | ~lart graphical file managers |
| 19:06.48 | Qwell | ender: :) |
| 19:06.53 | hardwire | no better way to do that than with a thumbnail viewing manager |
| 19:06.55 | harryvv | hardwire, which laptop? |
| 19:07.02 | hardwire | hardwire: Panasonic CF-73 |
| 19:07.16 | harryvv | is that the panasonic toughbook? |
| 19:07.32 | Qwell | hardwire: Just make sure you reconnect the fan, or the laptop will become a form of contraception |
| 19:07.33 | hardwire | you bet your harry iass it is |
| 19:07.34 | ender | hardwire: except a thumbnail viewer won't drag down the rest of your desktop software. |
| 19:07.51 | hardwire | ender: too early to comprehend |
| 19:07.59 | *** join/#asterisk huslage_ (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
| 19:08.07 | Qwell | hardwire: if it looks like explorer, acts like explorer, it'll probably crash like explorer |
| 19:08.16 | hardwire | yes |
| 19:08.32 | hardwire | but its easier than writing the filenames of 400 images down.. them moving them to a different dir |
| 19:09.01 | wunderkin | you have a problem remembering which of the girls are hot? |
| 19:09.02 | harryvv | hardwire, I have done alot of dell laptop repairs. All i can say is be carefull of the parts thay can be easily broken. one odd thing about the cpu modules for the dells, thay can require up to 35 bls of pushed down force to install into the module sockets. So I dont know about the tought book. |
| 19:09.19 | ender | hardwire: I'm not too fond of Naut trying to be the Do Everything (poorly) Tool. |
| 19:09.21 | hardwire | harryvv: no worries here.. I did benchwork fixing laptops |
| 19:09.27 | harryvv | k |
| 19:09.27 | Qwell | harryvv: my god, my PC fan takes like...a shitload of force |
| 19:09.33 | *** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de) |
| 19:09.33 | hardwire | ender: I don't use it as thus. so it seems to work fine for me |
| 19:09.35 | Qwell | heatsink/fan |
| 19:09.47 | harryvv | qwell yea..its scarry when putting that kind of force on a mobo ;) |
| 19:09.48 | Qwell | harryvv: I always have to get my wife to help me, heh |
| 19:10.05 | hardwire | harryvv: it has no fan on top of the processor |
| 19:10.13 | harryvv | interesting |
| 19:10.16 | Qwell | heatsink? |
| 19:10.44 | hardwire | harryvv: it has a fan attached to a sink |
| 19:10.51 | harryvv | the dells have some kind of tube that is copper welded to a small heat sink. tub leads to a fan on the back. |
| 19:11.13 | cripito | :P qwell my wife do it for me :P she is the one reparing laptops :))) i just look |
| 19:11.14 | hardwire | pretty much the same |
| 19:12.07 | Katty | mew. |
| 19:12.34 | *** join/#asterisk Tangent (n=Arc_Tang@82-40-187-54.cable.ubr06.croy.blueyonder.co.uk) |
| 19:12.59 | mutilator | well hell i see nothin wrong |
| 19:13.06 | mutilator | they look exactly the same |
| 19:13.17 | hardwire | http://catalog2.panasonic.com/webapp/wcs/stores/servlet/ModelDetail?displayTab=O&storeId=11201&catalogId=13051&itemId=65051&catGroupId=31954&modelNo=Toughbook-73&surfModel=Toughbook-73 |
| 19:13.30 | hardwire | mutilator: bad carriage return? |
| 19:13.34 | hardwire | in asterisk do show dialplan |
| 19:13.37 | *** join/#asterisk razu_ (n=razu@ip58.cab60.mus.starman.ee) |
| 19:13.37 | hardwire | and see if it shows up |
| 19:13.42 | harryvv | all our police cars use the Toughbook |
| 19:13.43 | ender | 'heatpipe' is what they're called |
| 19:13.54 | mutilator | see if what shows? |
| 19:13.58 | ender | pipes the heat away from the cpu to a cooling zone, usually w/ a fan near by |
| 19:13.59 | harryvv | thanks ender..i forgot the name. |
| 19:14.00 | hardwire | ender: shows how much I know |
| 19:14.15 | hardwire | mutilator: that did |
| 19:14.18 | ender | Shuttle PCs use them as well. |
| 19:14.24 | hardwire | they use toughbooks? |
| 19:14.33 | mutilator | yah it's in there |
| 19:14.35 | hardwire | my girlfriend hates it too |
| 19:14.37 | ender | hardwire: um, *thwap* |
| 19:14.39 | mutilator | else it wouldn't work when i change my callerid |
| 19:14.44 | hardwire | ok |
| 19:14.55 | hardwire | so I am going to take my P M 1.7ghz 1mb cache |
| 19:15.04 | hardwire | to P M 2.0Ghz 2mb cache |
| 19:15.10 | hardwire | that should be quite a step forward I hope |
| 19:15.27 | ender | the cache helps more than the ghz |
| 19:15.32 | hardwire | hmm |
| 19:15.36 | hardwire | my 1.7 ghas 2mb |
| 19:15.46 | hardwire | so I probably shouldn't care about an extra 300mhz |
| 19:15.56 | ender | I doubt you'll notice it that much. |
| 19:16.09 | hardwire | I would have a false feeling that would make me feel like I did |
| 19:16.19 | ender | yep |
| 19:16.24 | ender | subjective. |
| 19:16.30 | mutilator | yay! |
| 19:16.36 | hardwire | mutilator: ? |
| 19:16.47 | *** join/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 19:16.49 | mutilator | oh conversing with the gf |
| 19:16.55 | mutilator | Scott says: |
| 19:16.55 | mutilator | so what do you wanna do this weekend? |
| 19:16.55 | mutilator | Christina says: |
| 19:16.55 | mutilator | well, bowling, sex, i am not sure what else...what do u want to do |
| 19:17.02 | hardwire | yeh |
| 19:17.04 | *** part/#asterisk Pr0ph37 (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 19:17.06 | hardwire | I got some last night too |
| 19:17.10 | Qwell | sweet, bowling |
| 19:17.14 | hardwire | but we didn't go bowling |
| 19:17.21 | hardwire | I really feel like you are getting a better deal |
| 19:17.24 | Qwell | hardwire: :( |
| 19:17.26 | mutilator | i sure am |
| 19:17.46 | hardwire | ender: my price on the CPU is $304.28 |
| 19:17.52 | hardwire | thats 1 buck per mhz |
| 19:18.00 | hardwire | that I will gain |
| 19:18.04 | hardwire | don't think I will bite |
| 19:18.53 | hardwire | so I have a Pentum M 735 |
| 19:18.58 | hardwire | I want to go to a 760 |
| 19:19.03 | hardwire | I wonder what the heat difference is |
| 19:19.14 | hardwire | the latest toughbook in the same shell/mainboard as I have is at the 760 |
| 19:19.24 | hardwire | but they replaced the radeon w/ intel IGE |
| 19:19.38 | hardwire | err |
| 19:19.45 | hardwire | I have the term wrong.. but no GL :) |
| 19:19.54 | mutilator | argh |
| 19:20.02 | mutilator | i r pissed at this cisco box |
| 19:20.07 | mutilator | glad to be gettin rid of it |
| 19:20.29 | hardwire | if I stay at a 400mhz buss.. like IO probably should.. I can go to a Pentium 4-M 2.6 |
| 19:20.29 | mutilator | hopefully soon |
| 19:20.31 | ender | hardwire: wow, $300 for a measly 300mhz increase? totally not worth it. |
| 19:20.40 | hardwire | now whats the diff inbetween P M and P4 M ? |
| 19:20.46 | hardwire | do I all of a sudden have HT? |
| 19:20.54 | ender | hardwire: hrm, a typo? |
| 19:20.56 | hardwire | that may be cool |
| 19:21.06 | hardwire | ender: no.. I have seen tons of p4 mobile laptops |
| 19:21.09 | ender | hardwire: because Pentium M is a different class of CPU from a Pentium 4. |
| 19:21.11 | hardwire | vs p mobile |
| 19:21.16 | hardwire | ender: I agree |
| 19:21.25 | ender | hardwire: however, there are Pentium 4 processors found in laptops. |
| 19:21.29 | hardwire | but its here.. in the same FCPGA package. |
| 19:21.31 | ender | they are not as fast/good as Pentium M |
| 19:21.37 | hardwire | ender: really? |
| 19:21.43 | mutilator | s/laptop/portable desktop/ |
| 19:21.45 | hardwire | I thought an HT processor in a laptop would be outstandingly fun |
| 19:21.47 | ender | Pentium M usually has more cache |
| 19:21.51 | hardwire | oh |
| 19:21.54 | hardwire | yup |
| 19:22.01 | hardwire | 512 on the p4 mobile I am looking at |
| 19:22.01 | ender | mutilator: prior to the Pentium M there were Pentium 4 laptops. |
| 19:22.08 | hardwire | which is also $578.00 |
| 19:22.17 | ender | hardwire: P M usually has 1 if not 2 megs of cache. Makes a big difference. |
| 19:22.35 | ender | much snappier w/out the super high ghz heat/power penalty. |
| 19:22.42 | hardwire | ender: I believe they are as Intel describes "Mobile Pentium 4 Processor - M" because they are the Micro FCPGA package |
| 19:23.30 | hardwire | hah |
| 19:23.34 | hardwire | jesus |
| 19:23.39 | hardwire | this site is riddled with weirdness. |
| 19:23.47 | gandhijee | pentiumM is a P3 front end with a P4 backend tacked on to it |
| 19:23.49 | ender | Intel sucks for acronyms. |
| 19:23.50 | hardwire | most of these processors are 478-pin |
| 19:23.53 | hardwire | some are 479-pin |
| 19:26.09 | mutilator | anyoen ever setup voip and pots dial-peers on a cisco before? |
| 19:27.09 | wunderkin | justinu: you won a million dollars! wire me a million and a half and you get the mula! you were the closest on the price is right :D |
| 19:27.36 | harryvv | mutilator ask on #cisco on efnet |
| 19:29.06 | wunderkin | justinu: it was set to extended frame instaed of extended |
| 19:29.07 | wunderkin | er |
| 19:29.56 | *** join/#asterisk jwig (n=Joe@cherishbound2.dsl.xmission.com) |
| 19:35.33 | pc2 | harryvv - #cisco here works better. |
| 19:35.46 | Qwell | a fax is instant, right? |
| 19:35.53 | *** join/#asterisk benno2 (n=benno2@host75-45.pool8252.interbusiness.it) |
| 19:35.57 | wunderkin | superframe instead of extended super f rame maybe? s omething like that |
| 19:36.05 | Qwell | ie; if my fax machine says it was sent, that means the remote end received and printed it, right? |
| 19:36.20 | Qwell | (assuming the remote fax isn't out of paper) |
| 19:36.29 | FuriousGeorge | can someone recomend an analog handset to me? im just looking for a few programable buttons for things like call parking |
| 19:36.45 | Qwell | FuriousGeorge: hit radio shack, buy a cheapo $5 phone, heh |
| 19:36.48 | bawks | what happened to pbxfreeware.org? |
| 19:36.54 | Katty | FuriousGeorge: mew. |
| 19:36.57 | KranZ | Qwell: it also means some1 picked it up out of the tray and gave it to the intended person |
| 19:37.07 | Qwell | KranZ: excellent |
| 19:37.10 | KranZ | heh |
| 19:37.37 | fordvoice | ? |
| 19:37.37 | FuriousGeorge | hey Katty |
| 19:37.50 | fordvoice | What is the easiest way to program a Cisco ata 186 |
| 19:37.50 | *** join/#asterisk shimi (n=shimi@unaffiliated/shimi) |
| 19:38.08 | fordvoice | and how do you access it via the web |
| 19:38.12 | FuriousGeorge | Qwell: you cant even go into radio shack for five dollars |
| 19:38.22 | Qwell | FuriousGeorge: walk? :p |
| 19:38.32 | Qwell | oh, into...nm |
| 19:38.52 | FuriousGeorge | Qwell: maybe its just around here but their rj-45 terminations (heads) are like 8 bucks |
| 19:39.22 | Katty | ouch |
| 19:39.22 | FuriousGeorge | you cant get a dvi cable for under 120 |
| 19:39.24 | Katty | that's spensive. |
| 19:39.39 | *** join/#asterisk earthsound (n=webmonke@138.26.35.115) |
| 19:39.49 | cpatry | Katty: like i hate diamonds? :) |
| 19:40.17 | Katty | cpatry: mrow? |
| 19:40.19 | FuriousGeorge | granted the heads have a separate piece you feed the stuff into, but radioshack cat5 spools dont fit into newegg generic heads as a result |
| 19:40.34 | FuriousGeorge | ~beats radioshack |
| 19:40.38 | shimi | Hi all. I am having a problem, I am not sure if it's with Asterisk or not, but I am sure you guys can know :) I have setup an Asterisk machine using Asterisk@Home. It all works well with Xtensoftphone that I used. Now I've purchased several units of Grandstream GXP-2000 Enterprise phone. I connected it to my LAN, it took an IP address, and using the web interface I configured an account for a SIP extension on the asterisk machine. On the Flash Operat |
| 19:40.38 | shimi | or Panel, I can see the extension activity (i.e. I see that it dials, that it's busy, etc). The phone also makes busy sounds, and a dialtone, etc. However, when I am supposed to hear sounds (for instance, calling 8[ext] which is conference, where I should be getting "You're currently the only person in that conference"), I get a complete silence. Any hints? |
| 19:40.43 | Katty | cpatry: that did not parse. |
| 19:40.44 | bkw_ | RUDE RUDE RUDE |
| 19:40.46 | *** join/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
| 19:40.52 | Qwell | bkw_: he said hi |
| 19:41.06 | FuriousGeorge | i dont know if i should try to answer or grade it |
| 19:41.12 | bkw_ | haha |
| 19:41.16 | harryvv | radioshack was bought out by circuit city here in canada and changed to thesource |
| 19:41.18 | bawks | what happened to pbxfreeware.org? |
| 19:41.26 | Qwell | bawks: yell at bkw_ |
| 19:41.40 | FuriousGeorge | i remember the wonders of my tandy 8086 |
| 19:42.00 | harryvv | pbxfreeware is down |
| 19:42.16 | *** part/#asterisk Johnsie (n=john@acs-24-154-53-217.zoominternet.net) |
| 19:42.20 | *** join/#asterisk AsterNov (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk) |
| 19:42.59 | shimi | pbxfreeware.org loads for me |
| 19:43.04 | pc2 | FuriousGeorge - staples has an at&t trimline one free after mail in rebate starting the 8th :) |
| 19:43.29 | pc2 | pbxfreeware.org is up. It is hosted by cogentco, and you're probably singlehomed to level3.net |
| 19:43.34 | pc2 | see slashdot.org for the whole fiasco. |
| 19:43.47 | Ash | or the fun thread on nanog |
| 19:43.50 | bkw_ | there we go |
| 19:44.01 | bawks | working now |
| 19:44.23 | pc2 | Ash - url? |
| 19:44.35 | Ash | pc2: erm, google for 'nanog' and you will probably find list archives |
| 19:44.46 | Ash | I read it via NNTP from gmane.org |
| 19:44.48 | Katty | http://cgi.ebay.com/GORGEOUS-PURPLE-LACQUER-Piccolo-perfect-for-students_W0QQitemZ7355990913QQcategoryZ16229QQrdZ1QQcmdZViewItem <- want for birthday. |
| 19:44.52 | *** part/#asterisk jwig (n=Joe@cherishbound2.dsl.xmission.com) |
| 19:45.18 | shimi | anyone know anything about my problem? :) |
| 19:46.00 | Qwell | Katty: that title looks very bad |
| 19:46.11 | FuriousGeorge | pc2: whats this: http://www.thetwistergroup.com/store/customer/product.php?productid=BT110M%20L01277&source=fr |
| 19:46.38 | harryvv | with telus on strike the quality of service is sucking :) |
| 19:47.08 | pc2 | FuriousGeorge - a phone |
| 19:47.14 | Katty | Qwell: but i want a piccolo |
| 19:47.27 | Qwell | Katty: It makes a lot more sense once you click the link |
| 19:47.30 | pc2 | FuriousGeorge - an ugly one at that. |
| 19:47.32 | Qwell | I was almost afraid to though... |
| 19:47.45 | mutilator | argh |
| 19:47.51 | mutilator | this is mehhhhh |
| 19:47.57 | mutilator | shoot me in teh head |
| 19:48.54 | Katty | http://i16.ebayimg.com/04/i/04/c4/c7/ba_1_b.JPG <- also nice |
| 19:49.30 | Qwell | I like the purple one better |
| 19:49.36 | Katty | me too |
| 19:49.36 | Qwell | Did I just say that out loud? |
| 19:49.43 | Katty | purple is pretty |
| 19:50.24 | Katty | http://www.saletime.net/piccolo_purple_01.jpg |
| 19:50.39 | tzanger | looks like part of a clarinet |
| 19:51.00 | tzanger | Qwell: so long as you're not talking about a dildo it's all good |
| 19:51.30 | Katty | tzanger: it's a piccolo...a miniture flute |
| 19:51.51 | Katty | tzanger: the case is the side of your forearm |
| 19:51.55 | FuriousGeorge | pc2: im looking at that GE trimline phone, too |
| 19:52.31 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:53.43 | *** join/#asterisk jensdavidsen (n=jens@0x535ff4ce.vjnxx2.adsl-dhcp.tele.dk) |
| 19:53.59 | *** part/#asterisk jensdavidsen (n=jens@0x535ff4ce.vjnxx2.adsl-dhcp.tele.dk) |
| 19:59.32 | *** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net) |
| 20:02.59 | *** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca) |
| 20:05.34 | *** join/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net) |
| 20:07.13 | obsidian-studios | when deciding on a phone to buy, say there are 4 actual phone lines, would one need a 5 line phone? 4 physical lines plus 1 line for the phone itself? |
| 20:07.26 | Qwell | obsidian-studios: huh? |
| 20:07.27 | *** join/#asterisk loick (n=loick@APuteaux-151-1-15-115.w82-120.abo.wanadoo.fr) |
| 20:07.35 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 20:08.05 | obsidian-studios | Qwell: just trying to figure how to show others that someone is on a physical line, while still having an extension or line for inter company communication |
| 20:09.01 | FuriousGeorge | you know what would be kinda cool, yet somewhat out of left field. people have started using network cameras with two way audio for ip surveilance. they just get an ip on a network and send mjpeg or mpeg4 to whoever |
| 20:09.14 | FuriousGeorge | someone could make an ipcamera channel |
| 20:09.19 | FuriousGeorge | not me :| |
| 20:10.35 | obsidian-studios | Qwell: it's like this, with a normal pbx you have a button and a light for a line, I can pick up the phone and dial an extension to ring someone else. Or hit a button a use an outside line. So when using VOIP phones, just trying to figure out how to replicate that |
| 20:10.47 | Qwell | obsidian-studios: Thats a key system |
| 20:10.48 | FuriousGeorge | i think its a great idea |
| 20:11.08 | justinu | there are shared line appearances that may do what you want |
| 20:11.17 | justinu | i'm not sure if asterisk supports that |
| 20:11.49 | ender | obsidian-studios: thats on older lesser cool PBX systems, analog ones. |
| 20:11.55 | FuriousGeorge | obsidian-studios: i heard there was a sipura phone |
| 20:12.04 | FuriousGeorge | that mapped lights to parked calls |
| 20:12.11 | justinu | i have the sipura phone |
| 20:12.14 | ender | obsidian-studios: digital PBXs don't use that, you just have a line or 3 you can use for internal or external calling. |
| 20:12.19 | FuriousGeorge | but this is a new way of thinking about calling |
| 20:12.32 | obsidian-studios | I am just trying to figure out how to show which lines are in use on multiple phones. while also showing if that person is just on their person extension or using a physical line |
| 20:12.36 | ender | obsidian-studios: things will be much easier for your users if you break them of hte idea that line on their phone = POTS line to the building. |
| 20:12.39 | FuriousGeorge | if you have voice over ip and all your channels are busy, you could start making voip calls |
| 20:12.42 | FuriousGeorge | quite easily |
| 20:12.49 | ender | obsidian-studios: why? |
| 20:12.51 | FuriousGeorge | so you dont have a fixed amount of lines |
| 20:12.54 | justinu | obs: use FOP |
| 20:13.04 | Qwell | I could just imagine using a key system with multiple T1s |
| 20:13.17 | Qwell | 48 LEDs on your phone...that would suck |
| 20:13.27 | justinu | heh |
| 20:13.30 | FuriousGeorge | obsidian-studios: also, some sip clients, softphones at least, support contacts you could click on to send a message to or call or whatever |
| 20:13.34 | ender | lol. |
| 20:13.36 | obsidian-studios | Qwell: yeah, that's the extreme |
| 20:13.47 | justinu | the attendant consoles often have a lot of line appearances |
| 20:13.55 | ender | poor receptionist has 4 blocks of blinking lights. |
| 20:13.55 | justinu | even on PBXs |
| 20:14.33 | obsidian-studios | I am just thinking how VOIP phones show others what lines are in use or not. Not a big deal when grouping lines, and using a outbound group. But if someone wants to use a specific line, a way to show them before they pickup the phone. Someone else is already using tha tline |
| 20:14.41 | FuriousGeorge | pc2: i picked up 3 black att trimline shipped for 37.50 or so |
| 20:14.50 | ender | obsidian-studios: um, no. |
| 20:14.59 | ender | obsidian-studios: each phone 'line' is individual and indipendant. |
| 20:15.10 | ender | obsidian-studios: phone A's line 1 is completely different from phone B's line 1 |
| 20:15.31 | ender | obsidian-studios: w/ Voip (and w/ PRI and such) there isn't really 'specific lines' |
| 20:15.32 | obsidian-studios | ender: yeah, I get that, just trying to see if there are ways around that or not |
| 20:15.45 | FuriousGeorge | obsidian-studios: wouldnt you rather think about it this way: i never have busy lines? can you guys get business broadband to make sip or iax calls. probably cheaper anyway |
| 20:15.48 | ender | obsidian-studios: why would you want to 'get around' that? Advancements in telecom were made for a reason. |
| 20:16.05 | obsidian-studios | ender: like when a call comes in, hard for them to know what line it came in on, unless mapped to a particular channel on a phone with more than one |
| 20:16.11 | brad_mssw | obsidian-studios: you can do individual contexts for things, etc ... and make it work like that, but it'd be a PITA |
| 20:16.27 | FuriousGeorge | obsidian-studios: map you parked calls to extensions 1-9 so you can say "so and so is parked on line 1" then get the sipura phone that supports it |
| 20:16.29 | ender | obsidian-studios: w/ VOIP it didn't come in on a 'line'. It was bits in bandwidth. |
| 20:16.34 | brad_mssw | obsidian-studios: as for checking if a line is available ... you can always use the Asterisk Flash Control Panel thing |
| 20:16.41 | justinu | FOP |
| 20:16.47 | bjohnson | obsidian-studios: why do you want to know which line it came in on? |
| 20:17.07 | ender | obsidian-studios: again, if you put a call on "Line 1" of Phone A, there is absolutely no way for Phone B to see that Line 1 caller. |
| 20:17.11 | bjohnson | obsidian-studios: you could modify the CID to show that on the phone's screen |
| 20:17.20 | ender | obsidian-studios: the call would have to be parked then picked up, or transfered. |
| 20:17.24 | justinu | maybe their receptionist answers differently depending on what trunk it comes in on |
| 20:17.42 | bjohnson | justinu: get her a multi line phone |
| 20:17.47 | obsidian-studios | bjohnson: a manager might not answer all calls, but if the receptionist does not answer he can, Where I guess the VOIP way is to ring recp x times, then ring manager? |
| 20:17.51 | justinu | yep, that's one way to do it |
| 20:18.07 | justinu | you can make one inbound DID ring multiple phones |
| 20:18.11 | bjohnson | justinu: then each 'line" on the phone is one business/answer .. but multiple DIDs could feed each 'line' |
| 20:18.12 | ender | obsidian-studios: you can route calls however you want. |
| 20:18.17 | justinu | whoever answers first gets the call |
| 20:18.19 | bjohnson | obsidian-studios: yes |
| 20:18.30 | bjohnson | obsidian-studios: or ring both |
| 20:18.31 | obsidian-studios | ok, it's just going to mess with people used to the old way |
| 20:18.36 | justinu | fuck em |
| 20:18.42 | justinu | adapt or get out |
| 20:18.43 | benno2 | anyone expert in SIP text messages ? |
| 20:18.47 | ender | damn FOPs demo is broke. |
| 20:18.49 | obsidian-studios | I have no problem with it, but people used to pbx that have been around for years will bitch :) |
| 20:18.50 | FuriousGeorge | obsidian-studios: you put your pri lines in a group, calls come in top down, you dial out bottom up |
| 20:19.04 | bjohnson | obsidian-studios: explain that their three line just became 40 .. and they don't make phones with enough buttons |
| 20:19.12 | FuriousGeorge | if thats busy, use something else to make the call |
| 20:19.13 | ender | obsidian-studios: tell them it's progress. SHow them that Caller ID works, called ID works, parking, dynamic con-call rooms, etc... |
| 20:19.19 | bjohnson | obsidian-studios: it's really more open/flexible this way |
| 20:19.19 | benno2 | I've seen the hitachi WIP-5000 supports them and I tried to send one via VOIP provider sipphone.com (I think they use asterisk) and it got delivered correctly |
| 20:19.28 | justinu | set the voicemail up to email them their messages |
| 20:19.34 | FuriousGeorge | obsidian-studios: try to explain to them why it will be better |
| 20:19.34 | obsidian-studios | bjohnson: so you can't daisy chain cisco expansion modules ;) |
| 20:19.35 | *** part/#asterisk T0aD (n=toad@epsylon.org) |
| 20:19.40 | file | benno2: they use SER |
| 20:19.42 | benno2 | but the size seems to limited to 64 chars. could this be the phone or is this a limitaton in the SIP protocol ? |
| 20:19.53 | bjohnson | obsidian-studios: just to re-iterate .. pbx of any age don't do that .. you're talking about a key system |
| 20:20.21 | bjohnson | obsidian-studios: why would you want to |
| 20:20.26 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa31w-142166094161.nb.aliant.net) |
| 20:20.31 | benno2 | anyway its pretty amazing to call via WLAN from the hitachi. voice quality is excellent and if you optimize the phone settings (ie scan only the wi-fi channels you have APs on) it works very well |
| 20:20.32 | bjohnson | obsidian-studios: what purpose does it actually serve? |
| 20:20.33 | obsidian-studios | bjohnson: sorry, still getting down all the terms |
| 20:21.03 | benno2 | file: thanks for the clarification |
| 20:21.08 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-32-45.neo.res.rr.com) |
| 20:21.09 | obsidian-studios | bjohnson: just allowing someone to know what lines are in use and not. And place calls out a specific line, not always a main or random line |
| 20:21.25 | *** join/#asterisk dos000 (n=dos000@70.29.202.8) |
| 20:21.33 | bjohnson | obsidian-studios: pbxs are made for bigger systems. Your talking about a key system feature because they limited how many lines could be handled |
| 20:21.34 | ender | obsidian-studios: again, w/ VOIP you don't have 'lines'. |
| 20:21.39 | ender | obsidian-studios: you have to stop thinking about that. |
| 20:21.50 | FuriousGeorge | obsidian-studios: are you ever gonna run into a situation where all your lines are used and your bandwidth is all used up? |
| 20:22.02 | bjohnson | obsidian-studios: but that in itself is one of the benefits of voip, te flexibility to handle everything from little to large systems |
| 20:22.02 | benno2 | ender: lines = network bandwidth / 80kbit :) |
| 20:22.10 | gandhijee | can i not take a FWD call and send it to one of my Zap channels?? |
| 20:22.12 | ender | benno2: not the way that obsidian-studios is thinking about it. |
| 20:22.14 | bjohnson | FuriousGeorge: yes |
| 20:22.14 | obsidian-studios | ender: I know there are not "lines" I am just trying to figure out when buying a phone how many channels it needs to have |
| 20:22.16 | Qwell | gandhijee: sure you can |
| 20:22.29 | ender | obsidian-studios: how many calls do you want a phone user to handle at once? |
| 20:22.29 | gandhijee | hmm well i guess i need to check my configs again |
| 20:22.40 | ender | obsidian-studios: I found most people can't really use more than 3 lines. |
| 20:23.02 | ender | obsidian-studios: thats one call in progress, another incoming, and a third for transferring perhaps. |
| 20:23.03 | FuriousGeorge | obsidian-studios: are you talking about buying analog phones to use with your pbx or some flavor of voip phone |
| 20:23.10 | bjohnson | obsidian-studios: what purpose does calling out a specifiv line serve? |
| 20:23.21 | bjohnson | obsidian-studios: or knowing if it's busy or not? |
| 20:23.34 | obsidian-studios | bjohnson: some people have main business lines and their personal direct lines |
| 20:23.34 | ender | <PROTECTED> |
| 20:23.43 | FuriousGeorge | back to my idea of making a network camera channel, did you guys know most network cameras that have two way audio transmit their audio in ulaw |
| 20:23.49 | ender | obsidian-studios: Line 1 on Phone A being in use has absolutely no effect on Line 1 of Phone B |
| 20:23.53 | malverian[work] | I get really bad echo with my SIP -> PRI gateway. |
| 20:24.12 | malverian[work] | It's basically unbearable unless I use agressive echo suppression. |
| 20:24.15 | FuriousGeorge | malverian[work]: speakerphone? |
| 20:24.16 | bjohnson | obsidian-studios: get them a 2 line phone |
| 20:24.18 | mutilator | today is password change day |
| 20:24.28 | mutilator | engineer quit so we gotta change everything |
| 20:24.34 | mutilator | teh sux |
| 20:24.37 | ender | mutilator: those are fun days. |
| 20:24.45 | Qwell | mutilator: if it ends in a number, just increment it |
| 20:24.48 | Qwell | he'll never figure that out |
| 20:24.49 | bjohnson | the next days are more fun |
| 20:24.54 | bjohnson | trying to remember the new passwords |
| 20:24.56 | ender | mutilator: so is the day 3 days from now when you get woke up at midnight to fix something and you can't recall what you changed the password to. |
| 20:24.57 | mutilator | heh |
| 20:25.02 | Qwell | bjohnson: When 90% of them are locked out? |
| 20:25.20 | mutilator | our sql password hasn't changed in the 2 yrs i've been here |
| 20:25.23 | mutilator | and we gotta change that |
| 20:25.33 | mutilator | so in my password list of things to update |
| 20:25.41 | malverian[work] | FuriousGeorge, No, SNOM handset. |
| 20:25.43 | mutilator | i gotta figure out everywhere i have that password |
| 20:25.44 | malverian[work] | FuriousGeorge, And other. |
| 20:25.49 | mutilator | scripts and whatnot |
| 20:26.13 | bjohnson | ender: I find most people can't properly use one line |
| 20:26.28 | obsidian-studios | bjohnson: I was partly trying to figure out why the Grandstream GPX-2000 has 11 lines/channels |
| 20:26.34 | mutilator | New Text Document (3).txt |
| 20:26.40 | mutilator | all my passwords |
| 20:26.44 | ender | bjohnson: heh, thats a different matter. |
| 20:26.47 | mutilator | very inconspicuous |
| 20:26.49 | bjohnson | obsidian-studios: for people with too much money and little understanding |
| 20:26.49 | Qwell | mutilator: nobody will EVER find that. :) |
| 20:26.54 | wunderkin | justinu: did you see my comment earlier? you were right about the framing, thanks |
| 20:27.02 | bjohnson | obsidian-studios: my amp goes up to 11 syndrome |
| 20:27.03 | benno2 | obsidian-studios: btw how does the GXP-2000 work for you ? |
| 20:27.05 | FuriousGeorge | malverian[work]: i thought that pri didnt echo or something, i have no experience with it, are you using zap channels for your handset? |
| 20:27.06 | obsidian-studios | bjohnson: they are a dirt cheap multi-line phone? |
| 20:27.09 | earthsound | has kram been in here today? |
| 20:27.10 | *** join/#asterisk struct2 (n=struct@h236053.upc-h.chello.nl) |
| 20:27.25 | ender | obsidian-studios: $80 2-line phones exist |
| 20:27.26 | justinu | wunderkin: glad you got it worked out |
| 20:27.37 | wunderkin | me too, they kept testing back and forth and never saw a problem |
| 20:27.43 | justinu | yeah, they're morons |
| 20:27.46 | ender | obsidian-studios: I highly recommned the slightly more expensive Polycom IP-301 3-line phones. |
| 20:27.49 | obsidian-studios | benno2: it doesn't, still in the planning designing stage, not at purchasing just yet, trying to figure out how many phones are needed |
| 20:28.04 | justinu | the sipura 841 would be awesome if it wasn't for the buttons kinda sucking |
| 20:28.07 | sleepy_one | T100p T1 card with TA750 channel bank echoes bad :-( |
| 20:28.07 | ender | obsidian-studios: much better audio quality, hardware quality, and configurability. |
| 20:28.07 | struct2 | Hello, i have connected a Dect phone With ISDN DECT Accesspoint to my OCTOBRI card in asterisk |
| 20:28.08 | justinu | otherwise it's a nice phone |
| 20:28.10 | bjohnson | approx one per person |
| 20:28.18 | ender | justinu: and the audio quality sucks ass. |
| 20:28.19 | struct2 | 3 way call works nice, except the RETURN CALL doesn't work |
| 20:28.22 | obsidian-studios | the GXP-2000 is $83.95 http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-43410781952.htm |
| 20:28.26 | justinu | seems ok to me |
| 20:28.31 | benno2 | obsidian-studios: I made a few installs with a snom220 as the mainphone and budgetone 101 for guests (hotel) , works ok |
| 20:28.33 | struct2 | anyone an idea how to fix this |
| 20:28.36 | sleepy_one | I believe it is the TA750 channel bank that's the problem |
| 20:28.38 | ender | justinu: compare it to a Polycom. |
| 20:28.45 | justinu | gxp2000 is kinda cheesy, i have one here |
| 20:28.52 | justinu | ender: you get what you pay for... |
| 20:28.56 | FuriousGeorge | obsidian-studios: the number of "lines" voip phones have is a somewhat arbitrary concept. can a person have 17 quality conversations concurrently? beyond 5 it just gets silly for most purposes i think |
| 20:29.00 | ender | justinu: exactly. |
| 20:29.00 | justinu | sales monkeys don't care |
| 20:29.01 | obsidian-studios | just trying to figure out the need or justification for multi-line phones |
| 20:29.15 | obsidian-studios | FuriousGeorge: logical way to look at it |
| 20:29.19 | benno2 | justinu: cheesy in what sense ? sweet or still a barbie-phone ? :) |
| 20:29.21 | ender | justinu: they would if customers complain about sound quality. |
| 20:29.42 | FuriousGeorge | obsidian-studios: i dont think there is anything preventing anyone from making a 17 line phone. i dont know if the protocol limits it like that somehow |
| 20:29.42 | ender | justinu: which I do whenver I call a sales person and it sounds like I'm talking to him through a sock stuffed into a tin can dangling from a window. |
| 20:29.54 | obsidian-studios | FuriousGeorge: key system way a boss in small call center might want to keep an eye on lines in use, but I guess with * there are more elegant ways like via web or etc |
| 20:30.05 | justinu | is it only the tx audio that sounds like shit? |
| 20:30.11 | FuriousGeorge | obsidian-studios: exactly |
| 20:30.19 | ender | justinu: tx and rx IMHO |
| 20:30.33 | FuriousGeorge | obsidian-studios: you could even get a software package that blinked lights AND told you the cid, i bet |
| 20:30.34 | justinu | hmm, rx seems ok... certainly doesn't sound any worse than 2.4ghz cordless phones |
| 20:30.44 | ender | justinu: but most notably record somethign in VM using a Sipura, then listen to it on a Polycom. Then record w/ a polycom and listen again. |
| 20:30.54 | justinu | how about aastra? |
| 20:31.00 | obsidian-studios | with multi line phones, do people just match an internal extension to all, like in a group. So it rings what ever one is available or is a channel on the phone best left for internal stuff? |
| 20:31.00 | ender | never used it |
| 20:31.05 | justinu | polycom's are fugly |
| 20:31.26 | obsidian-studios | FuriousGeorge: yes, gotta have blinking lights, and lots of them |
| 20:31.28 | ender | obsidian-studios: a single ext is mapped to both/all the lines usually. |
| 20:32.06 | ender | obsidian-studios: exceptions would be people that answer for differnet thigns. They may have one line dedicated to a particular call type (which they answer differently than their personal extension) |
| 20:32.31 | ender | justinu: I find them rather nicely constructed compared to sipuras. The feel of them is much nicer too. |
| 20:32.47 | justinu | benno: build quality/interface |
| 20:33.22 | justinu | it's straight out of china |
| 20:33.26 | ender | justinu: ever used a Polycom IP-501 ? |
| 20:33.33 | obsidian-studios | it can be very simple for the person using the VOIP or give them a manual when dialing out and etc :) |
| 20:33.35 | justinu | but it's a 4 line phone for under $100 |
| 20:33.51 | ender | obsidian-studios: very easy. |
| 20:33.56 | fiber0pti | a 501 is under $100? |
| 20:33.58 | ender | obsidian-studios: pick up h and set, dial 9+phone. |
| 20:34.00 | ender | fiber0pti: no. |
| 20:34.04 | fiber0pti | oh..whew.. |
| 20:34.05 | fiber0pti | haha |
| 20:34.11 | justinu | ender not yet, i'm trying to figure out if I want a ip501 or an aastra 480i |
| 20:34.15 | fiber0pti | just aquired 16 of them for a little over $100 a piece |
| 20:34.15 | ender | fiber0pti: it's around $150 I think. and worth it. |
| 20:34.24 | justinu | fiber0pti: no, i was speaking of the gxp-2000 |
| 20:34.27 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 20:34.29 | fiber0pti | gotcha |
| 20:34.33 | obsidian-studios | ender: 9? I started getting rid of that so people can just pick up the phone and dial |
| 20:34.40 | ender | obsidian-studios: sure, you can do that too. |
| 20:34.47 | fiber0pti | I should be doing a deployment in the next couple of weeks with 10 polycoms |
| 20:34.51 | ender | obsidian-studios: you can match on 10 or 11 digits and send it out. |
| 20:34.52 | kpettit | can anybody point me to some docs that show me how to get t.37 or t.38 going? |
| 20:34.54 | fiber0pti | interested to see how it goes |
| 20:34.55 | obsidian-studios | ender: I mean we are thinking in new ways right, 9 is old school |
| 20:35.14 | justinu | if you want early (overlapped) dialing, you need 9 |
| 20:35.26 | ender | obsidian-studios: yes and no. We use 9 just so that a outgoing phone number that just happens to have the first 4 digits of somebody's extension won't be sent to someboey's extension. |
| 20:35.36 | Qwell | justinu: or 1, and 11 digit local dialing |
| 20:35.43 | justinu | yes, which kinda sucks |
| 20:35.49 | kpettit | I'm doing fax over ulaw right now but I'm geting alot of 1k pdf files out of it. I think the sip/ulaw thing isn't working consistantly so I'd like to start doing t.37 or t.38 but can't find any docs on how to implement it |
| 20:35.49 | obsidian-studios | ender: ah, good point |
| 20:35.53 | ender | obsidian-studios: say that Fred has extension 2062, and I want to call 206-295-4266 |
| 20:36.15 | obsidian-studios | ender: * does wait for them to stop entering digits? just flys with first match |
| 20:36.21 | fiber0pti | make 'em dial a 1 first for the long distance |
| 20:36.27 | Qwell | fiber0pti: and local |
| 20:36.29 | ender | obsidian-studios: w/out implimenting timeouts on the phone digitmap, and forcing users to dial quickly or put up w/ long delays before their numbers are sent, I use a 9 to totally seperate those calls out. |
| 20:36.37 | ender | fiber0pti: can't really do that. |
| 20:36.42 | justinu | where do I get polycom ip501s for 150 bucks? |
| 20:36.50 | ender | fiber0pti: there are numbers you have to 10 digit dial that won't work w/ a 1 |
| 20:36.58 | Qwell | ender: such as? |
| 20:37.00 | ender | justinu: tried voipsupply? what are they there? |
| 20:37.04 | fiber0pti | justinu: http://www.tritechcoa.com/ has them for 168 |
| 20:37.09 | fiber0pti | I got lucky on ebay though.. |
| 20:37.11 | ender | Qwell: in my area, I have to 10 digit dial 425 area code. |
| 20:37.19 | justinu | ender: 199 |
| 20:37.21 | Qwell | ender: you can't 11 digit dial it? |
| 20:37.21 | ender | Qwell: but 1-425 sends me to a completely different area. |
| 20:37.24 | mutilator | 2High4Work - good cisco enable pass ya think? |
| 20:37.25 | mutilator | :P |
| 20:37.25 | ender | Qwell: nope. |
| 20:37.25 | Qwell | lame |
| 20:37.31 | justinu | the aastra 480i is also 199 |
| 20:37.48 | ender | justinu: hrm, we paid less than that. |
| 20:37.57 | ender | justinu: CDW has them too, but I think we have a good discount w/ CDW |
| 20:37.58 | Qwell | ender: where does it send you? |
| 20:37.58 | *** join/#asterisk gvag11 (n=g@ppp22-adsl-105.ath.forthnet.gr) |
| 20:38.01 | justinu | the ip301 is 135 |
| 20:38.20 | ender | Qwell: to an operator who says 'you do not need to dial a 1....' |
| 20:38.21 | sleepy_one | ip501 $169 @ http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-44253921024.htm |
| 20:38.28 | Qwell | ender: ... |
| 20:38.29 | justinu | thx |
| 20:38.33 | Qwell | ender: seriously? |
| 20:38.35 | ender | Qwell: yes. |
| 20:38.41 | Qwell | thats fucking retarded |
| 20:38.43 | Lathos42 | I got a quote from Dell for the IP501 in the Bundle with the PoE cable for $201 |
| 20:38.46 | ender | Qwell: I know. |
| 20:39.01 | sleepy_one | ip301 $113 @ atacomm |
| 20:39.04 | ender | Qwell: i'ts the way 'local long distance' and 'long long distance' works aroun dhere. |
| 20:39.09 | justinu | nice |
| 20:39.18 | Qwell | ender: is it just that one area code thats like that? |
| 20:39.25 | ender | Definately call a sales person if you are going to buy in quantity (50~). THey'll mark down the price. |
| 20:39.27 | Qwell | If so, or if its only a few, you could setup a rule |
| 20:39.39 | ender | Qwell: well, if you're in area code 425, then calling 206 is like that too (: |
| 20:39.41 | justinu | yeah, unfortunatly i'm not buying 50 |
| 20:39.59 | Qwell | _1NXXNXXXXXX and _1425NXXXXXX |
| 20:40.09 | justinu | i heard some ramblings about polycom saying they weren't supporting asterisk |
| 20:40.13 | Qwell | the first one would pass ${EXTEN}, the second could pass ${EXTEN:1} |
| 20:40.22 | ender | Qwell: theoretically we could map aroun dthat, but I just have too setups _9NXXN and _91NXXN |
| 20:40.36 | ender | justinu: Polycom corp doesn't support Asterisk software. |
| 20:40.48 | ender | justinu: Polycom phones work swimmingly w/ Asterisk software. |
| 20:41.14 | justinu | it's a bullshit attitude for a company to have, imo |
| 20:41.15 | wunderkin | swimmingly? they don't drown? |
| 20:41.20 | Qwell | ender: do you have to 11 digit dial other areas? |
| 20:41.22 | *** join/#asterisk Mother_ (n=Mother@93.Red-80-32-127.staticIP.rima-tde.net) |
| 20:41.36 | ender | Qwell: well yeah, for long long distance calling. |
| 20:41.38 | fiber0pti | aww.. you guys are starting to make me worry about my phone purchase |
| 20:41.40 | Qwell | heh, silly |
| 20:41.44 | justinu | why? |
| 20:41.55 | ender | justinu: well, Polycom has their own call software that they want you to use. |
| 20:42.03 | wunderkin | Lathos42: dell? polycom? wow.. |
| 20:42.08 | Mother_ | greetings |
| 20:42.09 | sleepy_one | lol yeah so does cisco |
| 20:42.11 | fiber0pti | ender: so they don't work that well with asterisk via sip? |
| 20:42.14 | ender | Qwell: yeah, I wish it was like cell phones, just 10 digit dial EVERYTHING. |
| 20:42.20 | justinu | no, swimingly = fine |
| 20:42.22 | ender | fiber0pti: no, they work perfectly. |
| 20:42.26 | Qwell | ender: 10 or 11, but not both |
| 20:42.28 | fiber0pti | ohhhhhhhhhh |
| 20:42.30 | Mother_ | is it possible to give a zap dialtone to someone from an IVR? |
| 20:42.38 | Qwell | Mother_: look into disa |
| 20:42.38 | justinu | dict.org guys |
| 20:42.40 | sleepy_one | yes |
| 20:42.43 | fiber0pti | whew.. got me all worked up over the weeks of research I did.. ;) |
| 20:42.46 | justinu | lol |
| 20:42.49 | Mother_ | i.e. pressing option x will give you a dialtone on another zap channel |
| 20:42.58 | Qwell | Mother_: show application disa |
| 20:43.07 | Qwell | dialin something authentication |
| 20:43.09 | Qwell | or something |
| 20:43.11 | Qwell | ~disa |
| 20:43.16 | jbot | it has been said that disa is direct inward system access. show application disa |
| 20:43.16 | justinu | direct inward system access |
| 20:43.16 | Mother_ | yes but I don't want disa |
| 20:43.27 | Qwell | Mother_: That does exactly what you want |
| 20:43.31 | Mother_ | I'm using callback files |
| 20:43.37 | FuriousGeorge | when party outside of * calls party inside, on one specific zap channel, the PoS handset causes an echo when far party speaks loud. if i set the rxgain in zapata.conf to -1.0 would that help keepem quiet? |
| 20:43.54 | FuriousGeorge | so i dont send an echo back |
| 20:44.00 | FuriousGeorge | (new handsets are in the mail) |
| 20:44.15 | Mother_ | so far I have it working that you call * on a zap line, you hang up, if your callerid is on a list, it will call you back and give you a dialtone out another zap channel |
| 20:44.33 | Qwell | Mother_: DISA could still work in that scenario |
| 20:44.38 | ender | Mother_: DISA provides you a dialtone. Authenticated or unauthenticated. HOw is that not what you want? |
| 20:44.42 | Mother_ | but what I wanted to do is to call back the caller, but instead of giving them the dialtone right away, dump them into an IVR |
| 20:44.57 | Qwell | still not seeing a reason to not use DISA... |
| 20:45.13 | Mother_ | will DISA work after the call is established then? |
| 20:45.15 | obsidian-studios | with remote offices is it best to setup a VPN and have the phones talk to * directly via VPN. Or have an * box at each location. VPN between * boxes using IAX to communicate instead of SIP or etc? |
| 20:45.16 | Qwell | of course |
| 20:45.19 | justinu | use a context that has DISA rules after the IVR parts |
| 20:45.21 | Mother_ | i.e. jump to extension n |
| 20:45.26 | Qwell | Mother_: thats exactly what DISA DOES |
| 20:45.27 | sleepy_one | exten => 666,1,DISA,no-password|localx |
| 20:45.27 | Mother_ | OK thanks will look at it |
| 20:45.30 | ender | hrm, when you DIal( out to something, and that somebody picks up the call, how do you move them somewhere? You dn't go to the next priority line until that call is finished no? |
| 20:45.47 | justinu | ender: i've been trying to figure out that myself |
| 20:45.48 | Qwell | ender: You could use a macro. |
| 20:45.53 | Qwell | not sure what you're trying to do though |
| 20:45.55 | justinu | i've been exploring the dial macros |
| 20:46.00 | Qwell | option m to dial, I think |
| 20:46.02 | shimi | anybody knows a reason why would an IP phone give silence although sounds (should) have been sent from Asterisk ? |
| 20:46.04 | ender | obsidian-studios: setup another asterisk box and use IAX trunking between them. |
| 20:46.05 | justinu | M(macro) |
| 20:46.10 | Qwell | shimi: NAT |
| 20:46.23 | ender | obsidian-studios: that way local people in remote office can call eachother locally an dnot go out the VPN and back taking up lots of bandwidth. |
| 20:46.26 | shimi | and if I tell you that they're on Layer2 ? :) |
| 20:46.37 | Qwell | shimi: everything is on layer 2 |
| 20:46.37 | obsidian-studios | ender: that's what I thought, so that's ideal and phone to * via wan only in cases where a * box can't be put on both ends |
| 20:46.47 | Qwell | but yeah, NAT or firewall |
| 20:46.53 | ender | obsidian-studios: yeah. |
| 20:46.53 | shimi | let me rephrase - there are no routers between them... |
| 20:47.02 | Qwell | shimi: could still be nat or firewall |
| 20:47.07 | justinu | then one of your phones is using STUN |
| 20:47.13 | Qwell | more then likely the latter |
| 20:47.15 | justinu | and it's telling the other phone that it's RTP stream is on an external address |
| 20:47.19 | justinu | turn off STUN |
| 20:47.31 | shimi | NAT between two machines connected to the same switch? |
| 20:47.36 | shimi | justinu, was that for me? |
| 20:47.36 | Qwell | shimi: sure |
| 20:47.41 | justinu | shimi: yes |
| 20:47.46 | sleepy_one | shimi, are you running iptables on your * server? |
| 20:47.56 | shimi | well, I am running A@H |
| 20:47.59 | Qwell | shimi: two boxes on a switch could be on entirely different LANs |
| 20:48.01 | shimi | default configuration |
| 20:48.05 | Mother_ | Qwell & justinu: thanks it looks like that will work, I didn't think DISA would work after a call outbound |
| 20:48.13 | shimi | well, there are no VLANs there, if that's the question. |
| 20:48.15 | Qwell | Mother_: its just an application |
| 20:48.20 | *** part/#asterisk Ash (i=aaron@outofband.org) |
| 20:48.21 | shimi | it is to be noted that with softphone, everything works well |
| 20:48.24 | Mother_ | yes OK |
| 20:48.25 | shimi | STUN, you say ? |
| 20:48.26 | *** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com) |
| 20:48.32 | justinu | it could be STUN |
| 20:48.40 | justinu | it could be a local firewall on the PBX |
| 20:48.55 | shimi | justinu, but that would stop the softphone too, no? |
| 20:48.56 | obsidian-studios | is a cable modems upload 256-384 inconsistent going to cut the mustard with 9 inbound lines from a VOIP provider, and provding 3-4 lines to a remote office? |
| 20:48.58 | sleepy_one | shimi, turn on sip debug or whatever and see if the phones are able to talk to * at all |
| 20:48.59 | shido6 | any perl http users? |
| 20:49.03 | sleepy_one | me |
| 20:49.19 | Qwell | obsidian-studios: no, not really |
| 20:49.21 | justinu | yeah, but the softphone (xlite) may be smart enough to figure out that since the other phone is on the same subnet, don't use an external address in the SDP |
| 20:49.23 | shido6 | embperl? |
| 20:49.24 | Qwell | obsidian-studios: assume 80k per call |
| 20:49.50 | Qwell | that'd give you 3-4, at best |
| 20:49.50 | shimi | oh, they do talk. I see in the Flash Operator Panel that they do. I even get busy signals, etc, etc. it's just that I don't hear sounds from asterisk. SIP by itself seems to be working great |
| 20:49.52 | obsidian-studios | Qwell: regardless of codec? |
| 20:50.02 | Qwell | obsidian-studios: no, that is with ulaw |
| 20:50.02 | justinu | yes, SIP is just signalling |
| 20:50.13 | justinu | it's your RTP that's not getting to the proper destination |
| 20:50.14 | Qwell | others will be less, but...probably not enough to do 13 calls |
| 20:50.17 | justinu | and STUN can affect that. |
| 20:50.24 | shimi | I am pretty sure I've marked STUN as off. Perhaps this is related "Outbound Proxy" ? |
| 20:50.31 | justinu | maybe |
| 20:50.32 | sleepy_one | shimi, try tcpdump or ethereal see what kind of packets you're getting |
| 20:50.47 | *** join/#asterisk snaky (n=snaky@217.172.19.68) |
| 20:50.54 | shimi | sleepy_one, OK |
| 20:50.54 | obsidian-studios | Qwell: yeah, pretty sure it will be ulaw, but in lew of a different line they might elect for different codecs. I do not think it's going to work either way and call quality will suffer |
| 20:51.04 | Qwell | obsidian-studios: definitely |
| 20:51.18 | shimi | but one last question. Am i supposed to set "Outbound proxy" if all my connections are to Asterisk on the same l2 segment ? |
| 20:51.30 | justinu | don't think so |
| 20:51.44 | shimi | that might be it |
| 20:51.47 | sleepy_one | I think you do |
| 20:51.48 | shimi | thanks... |
| 20:51.52 | sleepy_one | maybe |
| 20:51.57 | sleepy_one | won't hurt to try |
| 20:52.08 | shimi | no, I've already set it. I am asking if I did wrong |
| 20:52.12 | ender | obsidian-studios: you need more. Especially if you're going to be doing regular data traffic off that link. |
| 20:52.13 | obsidian-studios | Qwell: I am shooting for split T1 voice/data, where the data portion 768-1.54 is solely for VOIP inbound and outbound. Of course not using a VOIP provider, but a regular telco |
| 20:52.15 | justinu | anyone versed in dial macros? |
| 20:52.19 | shimi | I can't really check now, as the phone is at work... :> |
| 20:52.41 | obsidian-studios | ender: I really prefer data and voice not to use the same pipe, otherwise one has to do QoS |
| 20:52.46 | justinu | i can't seem to get the h extension to work inside of a dial macro |
| 20:52.58 | ender | obsidian-studios: is there a particular reason to use a VOIP providor instead of just a number of PRI lines? |
| 20:53.15 | ender | obsidian-studios: say get 5~10 PRI lines (to handle up to 10 inbound/outbound calls) ? |
| 20:53.24 | ender | obsidian-studios: use IAX2 to connect your remote offices? |
| 20:53.38 | sleepy_one | IAX2 rocks :-) |
| 20:53.43 | sleepy_one | and is free |
| 20:53.48 | sleepy_one | minus bandwidth |
| 20:54.09 | sleepy_one | PRI = 300 - 3000 / mo |
| 20:54.09 | obsidian-studios | ender: well they have a few lines with Vonage. I am making the case that the cost of Vonage over 9 or so lines puts them in T1 range, and that they should get lines from telco |
| 20:54.34 | shimi | I got a PRI for free |
| 20:54.34 | shimi | :) |
| 20:54.49 | sleepy_one | how do u get a PRI for free?? |
| 20:54.55 | ender | obsidian-studios: ah yes. |
| 20:54.57 | shimi | it's called "competition" |
| 20:55.27 | ender | obsidian-studios: and they're quality will improve using T1 voice lines. No lossy compressions and such. |
| 20:55.37 | sleepy_one | some providers require your firstborn + a 5 yr contract to give u a PRI |
| 20:56.11 | ender | obsidian-studios: you'd practically need a full T1 worth of bandwidth to handle 9 VOIP lines, if they're heavy use of the 9 lines. |
| 20:56.22 | Katty | what can i use to burn audio and video_TS folders/files that dvd shrink creates for me? |
| 20:56.26 | ender | obsidian-studios: IE most the time using 5~8 of them at the same time. |
| 20:56.36 | ender | Katty: k3b |
| 20:56.37 | Katty | nero doesn't support my dvd burner. |
| 20:56.45 | justinu | what if he uses GSM codec |
| 20:56.45 | ender | Katty: oh, you're windows. |
| 20:56.49 | shimi | well, we have (had) a monopoly on telephony up until a few months ago. Now there's competition. They want customers (bad). They gave me PRI with 50 DIDs for one year and calls cheaper than what I paid the previous company... |
| 20:56.50 | Katty | ender: yes, this is a dual boot |
| 20:57.02 | obsidian-studios | ender: however I am in FL, and BellSouth is just not starting to offer flat rate long distance for businesses. Other telcos are starting to fall in line, decreasing the justification for cheap VOIP for long distance and etc |
| 20:57.05 | Katty | Hmmhesays: mrow? |
| 20:57.16 | mmlj4 | what could cause voicemailmain to not see keypresses? -- No username but # key pressed. Using CID '2076' / -- Playing 'vm-password' (language 'en') / -- Incorrect password '' for user '2076' (context = <any>) |
| 20:57.18 | obsidian-studios | ender: so even a 768k T1 ain't going to do much |
| 20:57.32 | Beirdo | get a room |
| 20:57.34 | Katty | file[laptop]: recommendation plskthx |
| 20:57.38 | Beirdo | :) |
| 20:57.38 | Katty | Beirdo: oh hush. |
| 20:57.43 | Katty | Beirdo: you're getting annoying. |
| 20:57.48 | Beirdo | heh, I'm being silly |
| 20:58.04 | Beirdo | sorry |
| 20:58.24 | ender | Katty: obsidian-studios well, 768, COMPLETELY dedicated to voip may be able to handle 9 OK. |
| 20:58.44 | ender | Katty: http://www.slysoft.com/en/clonedvd.html |
| 20:59.09 | obsidian-studios | ender: well at that point it would mainly be providing VOIP service between locations, The remote location needs 3 lines all the time, and there might be calls in between locations, 9 would be a max going both ways |
| 20:59.15 | ender | Katty: much better DVD copying software. Only transcodes what it absolutely has to to get your DVD to shrink to the right desired size, easy to use, veyr well put together. |
| 20:59.21 | *** part/#asterisk snaky (n=snaky@217.172.19.68) |
| 20:59.28 | *** join/#asterisk ncjp (n=switch@61.206.115.5.user.ad.il24.net) |
| 20:59.33 | *** join/#asterisk switch (n=switch@61.206.115.5.user.ad.il24.net) |
| 20:59.49 | ender | obsidian-studios: once you start using Asterisk, IAX2 takes care of all your interoffice stuff. |
| 20:59.49 | shido6 | turn trunking on both ends with gsm and kill the torrents and p2p porn shares |
| 20:59.51 | Qwell | OMFG |
| 20:59.55 | Qwell | THESE PEOPLE ARE FUCKING MORONS |
| 20:59.57 | Katty | ender: but not free. |
| 20:59.57 | Qwell | okay, get this |
| 21:00.08 | ender | obsidian-studios: I'm talking about what you use at the main office to get to the POTS network. |
| 21:00.09 | Katty | ender: i don't need to copy dvds. |
| 21:00.11 | Qwell | new apt complex faxes a form to old apt complex |
| 21:00.12 | ender | Katty: well..... |
| 21:00.19 | Qwell | old apt complex fills out form, sends fax back to new apt complex |
| 21:00.19 | Katty | ender: i know plenty about video editing |
| 21:00.20 | obsidian-studios | ender: ideally, but in the event they do not go for two * boxes, just want to make sure there is the bandwidth |
| 21:00.28 | Qwell | old apt complex says new apt complex didn't get filled out form. |
| 21:00.29 | Katty | ender: i simply need software that will burn to a dvd and support my burner. heh |
| 21:00.38 | shimi | I wonder when "POTS" will be renamed to "PATS" :) |
| 21:00.40 | ender | Katty: boot to Linux and use k3b |
| 21:00.43 | Qwell | old apt complex needs new apt complex to refax form, because when they fax things, they don't have it anymore |
| 21:00.50 | Qwell | ^ ABSOLUTE BULLSHIT |
| 21:01.02 | Qwell | yeah, because the god damned fax machine eats the fucking paper |
| 21:01.07 | Qwell | </rant> |
| 21:01.15 | Katty | ender: my movies are on the windows partition. |
| 21:01.15 | ender | obsidian-studios: if you don't have two * systems, then you'll need VOIP accounts for each of the remote sites too. |
| 21:01.28 | Katty | ender: and i will need it in windows for backup purposes, etc. |
| 21:01.29 | ender | obsidian-studios: and beware, a lot of VOIP services are one VOIP line _per phone_. |
| 21:01.38 | Katty | ender: thank you for your suggestion, but i find it is not what i'm looking for. |
| 21:01.48 | obsidian-studios | ender: main office will get lines via T1, TE110P that will split voice and data in the box. Ideally the data portion will only be used by * for remote IAX2 stuff |
| 21:02.07 | ender | Katty: sorry, I don't really use Windows for much other than gaming and saving netflix movies to watch at a later time. |
| 21:02.13 | *** part/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
| 21:02.18 | ender | obsidian-studios: gotcha. |
| 21:02.28 | obsidian-studios | ender: trying to stay away from a VOIP provider, and use a Telco, and they provide VOIP service to office, not re-provide ;) |
| 21:02.40 | ender | obsidian-studios: well, yeah, VPN the phone traffic, w/ the VPN overhead you have to overestimate the bandwidth usages. |
| 21:03.02 | DeeJayTwo | ender: when you say split voice and data.. |
| 21:03.04 | DeeJayTwo | what is this data? |
| 21:03.15 | ender | obsidian-studios: and remote interoffice calling doubles your usage, goes out to home office, then back to remote office extension again. |
| 21:03.44 | ender | DeeJayTwo: I didn't say it, obsidian-studios did. He's using a Full T1 to fulfill office's data needs via a few channels as well as their voice needs via a few channels. |
| 21:03.55 | obsidian-studios | ender: which is why I was like getting service from a VOIP provider, then connecting remote office to them via VOIP is goign to use allot of bandwidth |
| 21:04.21 | ender | obsidian-studios: yeah, I wouldn't consider a VOIP providor. |
| 21:04.33 | obsidian-studios | DeeJayTwo: some telcos give you a T1 and a channel bank, where you can plug in a switch and servers to data, and phones to voice side |
| 21:04.55 | obsidian-studios | ender: just does not make sense, the cost savings are hardly there, and the chance of call quality sucking is way to high |
| 21:05.03 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:05.16 | sahafeez | i have a tdm40b and a 110p in a box |
| 21:05.17 | ender | obsidian-studios: yep. |
| 21:05.33 | sahafeez | in zapata.conf do they need to be in a 2 differnet context |
| 21:05.52 | ender | obsidian-studios: but, in places that provide fibre internet to the house for dirt cheap, it's a hell o fa lot cheaper to use a VOIP service providor than to run a new PRI/T1 line. |
| 21:06.30 | ender | sahafeez: depends, do you want to have the calls that come in via those cards go to different places in your dialplan? |
| 21:06.39 | Icemaann | i could have sworn I read this somewhere, but I dont recall. with 1.2 isnt there a new way to do "reload extensions" ? |
| 21:06.48 | ender | Icemaann: extensions reload |
| 21:06.52 | Icemaann | ahhh |
| 21:06.54 | Icemaann | thats it |
| 21:06.57 | Icemaann | thanks ender |
| 21:06.59 | ender | np |
| 21:07.01 | ender | veyr handy |
| 21:07.02 | obsidian-studios | ender: please man, I am trying to live my life with blinders on. The fact that people have fiber lines to their house that make my expensive T1 fall to it's knees. Well makes me want to get a rifle and climb a clock tower |
| 21:07.04 | sahafeez | all inbound will be from the pri - the fxs goes to a fax |
| 21:07.10 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 21:07.19 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 21:07.26 | sahafeez | so if i get a ani of the fax line i want it to go to the fxs and out to the fax |
| 21:07.30 | Icemaann | Fiber to the house.... that sounds quite nice |
| 21:07.39 | ender | obsidian-studios: lol. Think about my $80 cable internet link that is 8mb down 768K up. I could do a lot of phone calls on that. |
| 21:07.40 | justinu | obsidian-studios:lol |
| 21:08.32 | obsidian-studios | ender: yeah I got that cable thing to, I think I am at 7 though :(. However I would not think of using it to provide business level VOIP service between locations and get signal from a VOIP provider |
| 21:08.51 | obsidian-studios | ender: why I have a T1 and cable modem, damn uploading is $$$ |
| 21:09.08 | Icemaann | im on a ds3 at the office too, just for data even ;-) |
| 21:09.18 | ender | obsidian-studios: right. Cable isn't dedicated bandwidth enough and guarenteed low latency enough for business level calling. |
| 21:09.24 | shido6 | turn trunking on both ends with gsm and kill the torrents and p2p porn shares/query sleeoy_one |
| 21:09.51 | dersteer | lol |
| 21:10.09 | obsidian-studios | ender: that and cable and dsl are best effort services. A policy telcos love to use here in the South. They figure they are dumb inbred southerns, what do they know |
| 21:10.20 | *** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz) |
| 21:10.22 | justinu | shido6 lol |
| 21:10.26 | patpatnz | hi everyone |
| 21:10.59 | obsidian-studios | shido6: deff got to have QoS rules for streaming porn. God forbid someone's surfing or phone traffic makes me miss a hump |
| 21:11.56 | ender | ok, I need to go and get FOP working. |
| 21:12.33 | Icemaann | I had it working with *@home, but I didnt like *@home. so im starting over lol |
| 21:12.46 | obsidian-studios | ender: personally I am Daper Dan man ;) |
| 21:12.48 | Icemaann | i guess really it boils down to me disliking AMP |
| 21:13.07 | patpatnz | does anyone know how/if you can set the callerid for a H.323 call to be the RPID passed by a sip client? |
| 21:14.16 | sahafeez | ok, so in zapata.conf for both the pri and fxs... |
| 21:14.21 | sahafeez | i am a bit confused |
| 21:14.25 | ender | Does the live demo work for any of you: http://www.asternic.org/ |
| 21:14.26 | JerJer | don't use H.323 |
| 21:14.28 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-237.cybersurf.com) |
| 21:14.32 | sahafeez | do i have to [channels] |
| 21:14.51 | ender | sahafeez: you're passing the calls coming in on the fxs port directly to the fax machine right? |
| 21:14.56 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:15.01 | ender | sahafeez: or you want to that is? |
| 21:15.03 | patpatnz | JerJer, you mean you don't use it or I shouldn't? |
| 21:15.07 | sahafeez | there are no calls on the fxs |
| 21:15.16 | ender | sahafeez: what is the FXS for then? |
| 21:15.17 | sahafeez | all calls will be inbound on the PRI |
| 21:15.24 | harryvv | Does anyone know this guy? he was mentioned on asterisk.org and now his story is on cnn.com |
| 21:15.26 | harryvv | http://edition.cnn.com/2005/TECH/10/05/katrina.tech.response.ap/index.html |
| 21:15.32 | sahafeez | to plug the fax into |
| 21:15.49 | ender | sahafeez: um... do you want to recieve faxes? |
| 21:15.58 | harryvv | err sorry he was mentioned on voip-info.org |
| 21:16.14 | sahafeez | i want an inbound call on the pri with an id of our fax to pass out to the fxs port to the fax |
| 21:16.21 | ender | sahafeez: ok. |
| 21:16.30 | ender | sahafeez: the 'context' in zapata.conf is only for inbound calls, not outbound. |
| 21:16.48 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 21:17.02 | ender | sahafeez: so in extensions.conf, just mattch the fax ID and Dial the Zap FXS channel. |
| 21:17.19 | sahafeez | ok. dont i need to define the fxs card in zapata.conf |
| 21:17.30 | ender | sahafeez: well, yes you do. |
| 21:17.35 | sahafeez | ok. so then.. |
| 21:17.40 | ender | sahafeez: if you want to call the device on the fxs channel, it has to be there. |
| 21:18.08 | sahafeez | ok. so the issue i have is.. |
| 21:18.09 | sahafeez | one sec |
| 21:19.19 | sahafeez | http://pastebin.com/385532 |
| 21:19.59 | ender | sahafeez: how is your /etc/zaptel.conf configured? |
| 21:20.23 | sahafeez | span=1,1,0,esf,b8zs |
| 21:20.23 | sahafeez | bchan=1-12 |
| 21:20.24 | sahafeez | dchan=24 |
| 21:20.24 | sahafeez | loadzone=us |
| 21:20.24 | sahafeez | defaultzone=us |
| 21:20.26 | sahafeez | fxoks=1-2 |
| 21:20.29 | sahafeez | defaultzone=us |
| 21:20.31 | justinu | can asterisk playback raw mu-law pcm files? |
| 21:20.31 | sahafeez | loadzone=us |
| 21:21.24 | sahafeez | http://pastebin.com/385536 |
| 21:21.27 | ender | sahafeez: hrm, I thought that the second card 'channel' would start w/ the next number from the first card. |
| 21:21.43 | Mother_ | can anyone identify the manufacturer of this? http://www.avanzada7.com/en_wifi.htm |
| 21:21.57 | *** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com) |
| 21:21.59 | sahafeez | in zapata? |
| 21:22.13 | sahafeez | ahh.. |
| 21:22.35 | ender | maybe not though. |
| 21:22.39 | sahafeez | no |
| 21:22.49 | sahafeez | same error cept says channel 13 |
| 21:22.53 | ender | yeah |
| 21:23.09 | Katty | mew :< |
| 21:23.27 | *** part/#asterisk oej (n=Olle@apollo.webway.se) |
| 21:24.27 | ender | sahafeez: when you startup zaptel service, does it see both the cards right? |
| 21:24.46 | sahafeez | yes. |
| 21:24.48 | ender | sahafeez: you have to get that much working before you can move on to Asterisk. |
| 21:24.59 | ender | sahafeez: what is the otuput of zaptel servce (ztcfg) |
| 21:25.24 | sahafeez | Notice: Configuration file is /etc/zaptel.conf |
| 21:25.24 | sahafeez | line 7: Channel 1 already configured as 'Clear channel' at line 2 |
| 21:25.24 | sahafeez | line 7: Channel 2 already configured as 'Clear channel' at line 2 |
| 21:25.25 | sahafeez | hum |
| 21:25.45 | Qwell | fxoks=13-14 |
| 21:25.46 | Qwell | no? |
| 21:25.47 | *** join/#asterisk professorchen (n=mydejama@82-70-183-14.dsl.in-addr.zen.co.uk) |
| 21:25.50 | sahafeez | yes. think so |
| 21:25.54 | Qwell | erm, 25-26 |
| 21:26.13 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 21:26.21 | professorchen | where is res_data project, svn.asteriskdocs.org seems to be down |
| 21:26.31 | ender | sahafeez: yeah, you need to adjust your /etc/zaptel config. |
| 21:26.37 | sahafeez | cool. thanks. |
| 21:26.51 | ender | np |
| 21:27.26 | pifiu | why does thevoice.digium.com take so long to make my recordings? |
| 21:27.30 | pifiu | allison you're slacking! |
| 21:28.02 | obsidian-studios | ok, regardless of protocol, bandwidth is mainly determined by codec correct? Or does IAX have less overhead than SIP? |
| 21:28.35 | ender | obsidian-studios: IAX can do trunking and less overhead. |
| 21:28.37 | justinu | even so, it probably means nearly nothing compared to the RTP streams |
| 21:28.57 | ender | obsidian-studios: SIP is going to be individual streams each w/ their own overhead. |
| 21:29.18 | obsidian-studios | ender: wondering #'s to base the overhead on, like say 9 lines again via IAX? |
| 21:29.20 | ender | obsidian-studios: and SIP is mostly UDP fire and forget. IAX is going to be TCP IIRC. |
| 21:29.37 | ender | obsidian-studios: hrm, hard to say, I haven't done any testing myself. |
| 21:29.43 | JerJer | ender: um hell no |
| 21:29.48 | obsidian-studios | what does Vonage use between them and the device? SIP? |
| 21:29.49 | JerJer | iax is udp |
| 21:29.55 | ender | JerJer: oh whoops. |
| 21:29.56 | justinu | use etherreal to watch the iax packet sizes |
| 21:30.01 | justinu | for one call |
| 21:30.03 | JerJer | SIP uses an RTP encoded udp stream |
| 21:30.08 | justinu | then extrapolate from that |
| 21:30.11 | obsidian-studios | you can send SIP over TCP can't you? |
| 21:30.13 | sahafeez | got it all. thanks. |
| 21:30.19 | obsidian-studios | instead of UDP? |
| 21:30.24 | JerJer | SIP signaling messages can be TCP yes |
| 21:31.42 | harryvv | is this channel being logged? |
| 21:31.47 | Qwell | harryvv: somewhere |
| 21:31.51 | harryvv | need some info from it |
| 21:32.43 | netsurfer | harry - wassup? i been here most of the day |
| 21:32.44 | ender | harryvv: from how long ago? |
| 21:32.50 | netsurfer | and have a big buffer ;) |
| 21:32.51 | ender | harryvv: I log all my irc channels. |
| 21:33.05 | ender | harryvv: and I haven't disconnected from this channel in weeks. |
| 21:33.26 | harryvv | netsurfer, there is somone who was showing a asterisk managment panel for windows. I dont recall his nick. He looked to have some really good html and dev skills. |
| 21:33.43 | ender | harryvv: how long ago? |
| 21:33.53 | harryvv | He is/was here in the last 3 to 4 hours ago |
| 21:33.57 | netsurfer | well since ender thinks he's logging the world i'll leave it to him and drink my coffee |
| 21:34.00 | ender | do you recall the name? |
| 21:34.02 | harryvv | he showed a number html sites. |
| 21:34.04 | harryvv | no |
| 21:34.14 | ender | of the management tool |
| 21:34.16 | harryvv | of i would not be asking. |
| 21:34.17 | harryvv | yes |
| 21:34.21 | ender | or some text he said that I could grep for. |
| 21:34.35 | harryvv | http://voipgw.cripiland.com/ |
| 21:34.44 | ender | brb |
| 21:35.20 | Qwell | cripito |
| 21:35.21 | ender | cripito |
| 21:35.51 | ender | 13:49 ::: cripito!n=ncripito@67.96.197.99 has quit: |
| 21:35.56 | harryvv | k |
| 21:35.58 | harryvv | yea |
| 21:36.06 | harryvv | no way to contact him ;) |
| 21:36.07 | ender | right now it is 14:34 my time |
| 21:36.41 | ender | harryvv: his nick is registered, you can leave him a message w/ memoserv. |
| 21:36.47 | harryvv | I have thought more of a wisp setup |
| 21:36.58 | harryvv | okay |
| 21:37.02 | zoa | what were the urls he showed ? |
| 21:37.05 | ender | 11:19 <cripito> check this out http://www.cripiland.com/screenshots/manager1.jpg |
| 21:37.09 | ender | 11:20 <cripito> and this http://www.cripiland.com/screenshots/manager2.jpg |
| 21:37.12 | harryvv | That was one of them |
| 21:37.52 | zoa | those are not websites :) |
| 21:38.06 | harryvv | we know |
| 21:38.30 | Qwell | are there any operator panels like that...that aren't windows clients? heh |
| 21:38.36 | zoa | http://www.asteriskguru.com/tools/switchboard.php does something similar |
| 21:38.39 | zoa | but its also windows |
| 21:38.44 | zoa | we might make a linux version though |
| 21:38.58 | ender | python would be good |
| 21:38.58 | Qwell | zoa: port to html :p |
| 21:39.01 | ender | cross platform. |
| 21:39.08 | zoa | idefisk linux is almost ready |
| 21:39.09 | Qwell | hell, even something proprietary, like asp |
| 21:39.20 | Qwell | as long as linux or windows machines could use it :p |
| 21:40.08 | Dr_Ray | how about not flash oriented |
| 21:40.28 | zoa | yeah, i also dont like flash |
| 21:40.31 | harryvv | so how do you use memoserv |
| 21:40.35 | zoa | especially that behaviour on right click |
| 21:40.40 | Qwell | harryvv: msg memoserv help |
| 21:40.57 | Dr_Ray | flash plus mozilla equal death to my PC |
| 21:41.12 | ender | flash are teh dead |
| 21:41.20 | sahafeez | hum. ok. so i want to all inbound calls to be routed by what number was dailed. there will be no DID |
| 21:41.22 | Qwell | flash was never alive |
| 21:42.21 | Dr_Ray | you mean no hunt group |
| 21:42.52 | ender | sahafeez: what does your telco pass to you? |
| 21:43.20 | sahafeez | calling and asking them |
| 21:44.03 | harryvv | Thanks Qwell. never knew there was such a beast. BTW Mac do you know mac Dearmon? Seems his asterisk story has migrated to cnn.com |
| 21:44.27 | harryvv | I mean Qwell, Do you know him. |
| 21:44.35 | Qwell | no |
| 21:44.37 | *** join/#asterisk timecop (i=timecop@AnimeNfo.com) |
| 21:44.39 | Qwell | I don't know anybody |
| 21:44.52 | harryvv | :) |
| 21:45.08 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 21:45.21 | harryvv | I have made these SAT antennas for hamradio before. I suspect he is uplinking to some commercial sat with wifi. |
| 21:45.23 | harryvv | http://edition.cnn.com/2005/TECH/10/05/katrina.tech.response.ap/index.html |
| 21:45.37 | professorchen | where is the stuff previously at svn.asteriskdocs.org now hosted? |
| 21:45.40 | Hmmhesays | hrm my callfile doesn't want to put me into a meetme conference |
| 21:45.47 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
| 21:45.49 | Qwell | ~asteriskdocs |
| 21:45.56 | MuppetMaster | Hello. |
| 21:45.59 | MuppetMaster | Thoughts on this?: http://www.openpbx.org |
| 21:46.59 | X-Rob | MuppetMaster - join #openpbx |
| 21:47.07 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 21:47.12 | MuppetMaster | But why the fork? |
| 21:47.27 | X-Rob | It's on the wiki |
| 21:47.39 | Mother_ | the reason is on the wiki? |
| 21:47.44 | X-Rob | yea |
| 21:47.46 | FuriousGeorge | Qwell: what was that link you'd sent me for the surpluss parts |
| 21:47.47 | harryvv | http://edition.cnn.com/2005/TECH/10/03/mac.dearman/index.html |
| 21:47.55 | Mother_ | hmaky checking... |
| 21:47.57 | MuppetMaster | But why are you saying we should join? |
| 21:47.58 | paryl | i have a couple gxp-2000's here and i've been doing testing with them. i can't here myself talking while using the handset... is that a normal thing? |
| 21:48.10 | Qwell | FuriousGeorge: umm, http://www.redemtech.com/webstore/ |
| 21:48.12 | Qwell | something like that? |
| 21:48.19 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:48.27 | FuriousGeorge | thats the one |
| 21:48.47 | sahafeez | ok. i should be getting the last 4 of the number dialed. |
| 21:48.54 | MuppetMaster | And which wiki for the reason for OpenPBX.org? Link? |
| 21:49.06 | ender | sahafeez: sounds like how DIDs work. |
| 21:49.13 | ender | sahafeez: so then just match and route on XXXX |
| 21:49.27 | MuppetMaster | This one I guess: http://wiki.openpbx.org |
| 21:49.37 | MuppetMaster | Still, why? How many of the developers are really jumping ship? |
| 21:49.39 | harryvv | I need a did |
| 21:49.46 | X-Rob | paryl - yes, that's normal. |
| 21:50.01 | Hmmhesays | anyone else have trouble with callfiles and meetme? |
| 21:50.07 | professorchen | there doesn't seem to be anyone at asteriskdocs, I am quite new to IRC, is the something I haven't down right an @ appears before my name |
| 21:50.11 | X-Rob | MuppetMaster - it's not really jumping ship. It's a fork, that's all. |
| 21:50.27 | Katty | hmm |
| 21:50.35 | MuppetMaster | But doesn't that defocus a bit? |
| 21:50.48 | MuppetMaster | Would seem time spent by developer's on OpenPBX is not time spent on Asterisk. |
| 21:50.48 | X-Rob | A lot. |
| 21:50.54 | X-Rob | Yes indeed. |
| 21:50.58 | MuppetMaster | So, kind of like jumping ship. |
| 21:51.07 | Dr_Ray | well, it is their time to spend as they see fit |
| 21:51.14 | X-Rob | Bingo, Dr_Ray |
| 21:51.22 | *** join/#asterisk gandhijee (n=gandhije@host-66-202-34-162.spr.choiceone.net) |
| 21:51.32 | paryl | X-Rob: but's it's not the usual behavior for every other phone... is there a way to adjust that? |
| 21:51.45 | MuppetMaster | Indeed, but why is the fork really necessary? Key drivers? |
| 21:51.52 | MuppetMaster | What will OpenPBX provide? |
| 21:51.52 | tzanger | MuppetMaster: the probloem is that the people in openpbx feel they are hitting too many roadblocks to be useful |
| 21:52.00 | X-Rob | paryl - not at the moment, but they have said they'll look at it in future firmware revisions |
| 21:53.00 | MuppetMaster | Reading the roadmap: http://wiki.openpbx.org/tiki-index.php?page=RoadMap |
| 21:53.11 | MuppetMaster | So Mark Spencer is no longer the Asterisk community darling? |
| 21:53.26 | Dr_Ray | he is in my book |
| 21:53.39 | Dr_Ray | but I al so like the openpbx guys |
| 21:53.43 | ender | from the roadmap, it doesn't seem to be much o f afork, a fork would assume that all / most original code is used, and then diverges. |
| 21:53.53 | ender | it would almost seem like a completely new project. |
| 21:54.04 | MuppetMaster | Indeed, lots of talk about ripping out current code and capabilities. |
| 21:54.08 | Dr_Ray | it's a GPL thing, I think |
| 21:54.17 | MuppetMaster | Not really much of a fork, more of 2 steps back and maybe 3 forward? |
| 21:54.38 | MuppetMaster | Is this an issue with the Asterisk Business Edition and the recent Intel card support? |
| 21:54.38 | *** join/#asterisk websae (i=websae@207-118-128-43.dyn.centurytel.net) |
| 21:55.05 | Dr_Ray | I think it's an issue with new code not making it in to asterisk |
| 21:55.13 | Dr_Ray | but I dunno |
| 21:55.19 | MuppetMaster | Well, one does like stability in a telephony platform. |
| 21:55.38 | wunderkin | i also think it has to do with someone able to make money off of their code |
| 21:55.41 | Dr_Ray | I mean, if that is how they want to spend their time, good for them |
| 21:57.21 | X-Rob | There's nothing wrong with making money from GPL code. |
| 21:57.39 | Mother_ | indeed...unless you're a purist...eeew |
| 21:57.59 | MuppetMaster | X-Rob: Indeed, there is not. |
| 21:58.19 | Dr_Ray | I view asterisk at home as a bigger "cancer" |
| 21:58.25 | MuppetMaster | Would prefer to see a meeting of the minds on Asterisk in order to have the community benefit from all of the best and brightest. |
| 21:59.04 | wunderkin | i view it as a group effort, im doing what i can since i benefit from it |
| 21:59.15 | MuppetMaster | Benefit from? |
| 21:59.23 | wunderkin | the use of asterisk |
| 21:59.28 | wunderkin | in business |
| 21:59.32 | MuppetMaster | Yes, as do I. |
| 22:00.09 | MuppetMaster | And I agree. There are some great features listed on OpenPBX roadmap, but also a lot of 'ripping'. But would prefer to see those in Asterisk. |
| 22:00.18 | paryl | X-Rob: heh, apparenlty it has been fixed. i installed 1.0.12 and it seems to have fixed things quite nicely |
| 22:00.27 | paryl | w00t |
| 22:00.30 | *** part/#asterisk pc2 (n=pc@209.151.52.81) |
| 22:00.33 | Icemaann | how are you guys using asterisk at work? just curious in hearing some use cases. |
| 22:01.20 | pifiu | yeah but as shido6 said |
| 22:01.34 | MuppetMaster | I have used Asterisk for a long time as a SoHo system for working from home. In the office right now we use it to provide voicemail off of a 'legacy' PBX. Also working on some other apps. |
| 22:01.38 | Dr_Ray | I use zaptel to provide hotel extensions |
| 22:01.59 | MuppetMaster | I don't know what to make of this fork, not happy about it. |
| 22:01.59 | sleepy_one | T1 PRI <-> Digium T100p <-> * <-> cisco 7960s # with VoIP backup |
| 22:02.00 | Icemaann | we are looking at using it to add conference support |
| 22:02.20 | MuppetMaster | But, indeed, it is up to the community members to decided where time is spent... |
| 22:02.27 | Icemaann | we use a iwatsu phone system now, no conf support |
| 22:02.30 | MuppetMaster | decide that is |
| 22:02.32 | generalhan | sleepy_one: how many 7960s you using >? |
| 22:02.43 | Dr_Ray | Muppet - It's thier time to spend.. maybe something great will come of it.. |
| 22:03.01 | MuppetMaster | Maybe, but now there are two projects instead of one, neither will move as fast. |
| 22:03.09 | MuppetMaster | Or if one does, it will suffer from stability issues. |
| 22:03.16 | MuppetMaster | This is a setback near term |
| 22:03.23 | paryl | do you guys think the gxp-2000 is good enough for a large rollout, or should i look at better phones? |
| 22:03.28 | Dr_Ray | well, I don't beleive asterisk will be less stable because of it |
| 22:03.53 | MuppetMaster | I agree Asterisk will not be less stable, but it will not have as much manpower behind it so development may slow. |
| 22:04.16 | MuppetMaster | Whereas if OpenPBX wants to do all it states in the roadmap in any kind of timeframe will suffer stability issues. |
| 22:04.30 | MuppetMaster | So both will be hampered by the fracture. |
| 22:04.34 | Dr_Ray | I dunno, asterisk still gets new people |
| 22:04.37 | MuppetMaster | IMHO |
| 22:04.38 | obsidian-studios | MuppetMaster: what's up with this fork, is there a link so I can catch up? |
| 22:04.44 | MuppetMaster | http://wiki.openpbx.org/tiki-index.php?page=RoadMap |
| 22:04.57 | MuppetMaster | http://www.openpbx.org/ |
| 22:04.58 | obsidian-studios | MuppetMaster: ty |
| 22:05.03 | harryvv | ahh heck, iax.cc circuits are bussy now. guess all there truncks are used up. |
| 22:05.11 | *** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) |
| 22:05.11 | MuppetMaster | But some of those new users will go to OpenPBX.org as opposed to Asterisk. |
| 22:05.29 | MuppetMaster | This will be a resource drain on Asterisk no matter how you slice and dice it, if OpenPBX.org manages to get off the ground. |
| 22:05.37 | jlewis | is there any way in CLI or manager to verify which channels are in a zap group? |
| 22:06.18 | obsidian-studios | MuppetMaster: hmm, does not seem good |
| 22:06.46 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
| 22:06.48 | MuppetMaster | I do not see it as positive for Asterisk or the alternative, OpenPBX |
| 22:07.02 | MuppetMaster | Can't people just get along? |
| 22:07.18 | obsidian-studios | MuppetMaster: this is the result of code not making it into * |
| 22:07.22 | obsidian-studios | ? |
| 22:07.36 | Dr_Ray | should redhat, mandrake, debian, and slackware not formed their own linux distributions? |
| 22:07.37 | MuppetMaster | I guess so, people want to shove lots of features into Asterisk. |
| 22:07.47 | tzanger | obsidian-studios: no, this is a combination of factors |
| 22:07.53 | MuppetMaster | But, indeed there are arguments for pacing the features, as stability is now important for Asterisk. |
| 22:08.07 | MuppetMaster | Well, will watch this one closely. |
| 22:08.07 | obsidian-studios | tzanger: support for more than digium hardware? |
| 22:08.15 | tzanger | 1) patches that seem to be proven to work being flat-out ignored. 2) tightlipped development from asterisk 3) apparent lack of interest/involvemnt of the community by digium |
| 22:08.25 | tzanger | obsidian-studios: it already supports more than digium hardware |
| 22:08.26 | Mother_ | it is *very* important, the concept of 'rebooting your PBX' doesn't sit well with many people |
| 22:08.38 | MuppetMaster | Posted a link here: http://forums.digium.com/viewtopic.php?t=1785 to see what more of the community has to say. |
| 22:08.39 | obsidian-studios | tzanger: ah, good old all around controversy on all fronts |
| 22:09.21 | MuppetMaster | Digium may be moving more towards commericialization, just like Mambo did recently which alientated their users. |
| 22:09.38 | MuppetMaster | Not saying that Digium did not bring some of it on themselves. Still hate to see it. |
| 22:09.50 | MuppetMaster | Well, late in Barcelona. Time for bed. G'night. |
| 22:09.55 | tzanger | I have zero problem with ABE |
| 22:10.04 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193) |
| 22:10.06 | websae | I have a question for everyone....I have an asterisk server I am trying to connect to...I can make calls out, but I can't hear the person, they can hear me...I am behind a router, but I have port 5060 forwarded to my Sipuara 841 Hard Phone |
| 22:10.10 | X-Rob | MuppetMaster - no-one reads the forums. |
| 22:10.15 | X-Rob | ...and he's gone. |
| 22:10.15 | websae | anyone have any ideas? |
| 22:10.16 | websae | :) |
| 22:10.29 | tzanger | websae: you need your RTP ports forwarded too |
| 22:10.34 | tzanger | 5060 is just SIP signaling |
| 22:10.40 | tzanger | there *should* be data on this in the wiki |
| 22:10.52 | websae | TP? |
| 22:10.53 | X-Rob | ~nat |
| 22:10.54 | jbot | methinks nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 22:10.56 | websae | *RTP |
| 22:11.00 | websae | i have nat=yes |
| 22:11.03 | Qwell | ~rtp |
| 22:11.04 | jbot | methinks rtp is The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 22:11.04 | adelas | any recomendation for a good SIP gateway? |
| 22:11.05 | websae | and qualify=no |
| 22:11.18 | Icemaann | adelas: SER? |
| 22:11.35 | adelas | u mean the brand "ser" |
| 22:11.36 | adelas | ? |
| 22:11.42 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 22:11.49 | Icemaann | http://www.iptel.org/ser/ |
| 22:11.51 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:11.52 | tzanger | websae: that's only part of it, you still need to forward the RTP ports |
| 22:11.59 | websae | on my netgear router |
| 22:12.05 | websae | there is no option for that |
| 22:12.10 | Icemaann | adelas: never used it myself, but they have some big claims |
| 22:12.18 | websae | you just select your ports your want to forward |
| 22:12.27 | websae | doesn'ta sk about UDP or RTP or anything like that |
| 22:12.29 | adelas | hey thats no good :P\ |
| 22:12.38 | adelas | i meant SIP fxo gatway |
| 22:12.48 | Icemaann | adelas: ahhh |
| 22:14.42 | websae | any ideas? |
| 22:14.49 | websae | i have tried just about everything |
| 22:14.53 | Ariel_ | Icemaann, which one do you know? |
| 22:15.11 | Qwell | websae: You need a better router then... |
| 22:15.56 | generalhan | anyone know an sntp server on the internet that i can point my 7960s to ? |
| 22:17.12 | Ariel_ | time.apple.com |
| 22:17.33 | Ariel_ | I have had problem with the small netgear routers. |
| 22:17.35 | gandhijee | anyone know a working FWD that ican try? |
| 22:17.45 | toddf | 613 |
| 22:17.47 | gandhijee | everytime i try 393612 i get a busy signal |
| 22:17.57 | gandhijee | that explains it |
| 22:18.08 | Icemaann | Ariel_: i use sipura ATA devices. not sure if thats what your looking for |
| 22:18.16 | harryvv | Ariel, how many areas in your local dont have high speed internet access? |
| 22:18.29 | Ariel_ | Icemaann, I use them but there only one fxo and one fxs |
| 22:19.00 | Ariel_ | harryvv, just a few. Mostly outside the 20 mile coverage from the center of City of Miami |
| 22:19.06 | Ariel_ | I am almost there |
| 22:19.44 | *** join/#asterisk websae2k (i=websae@207-118-134-96.dyn.centurytel.net) |
| 22:20.06 | harryvv | Looking into starting Wisp voip/internet services instead of the traditional isp model. |
| 22:20.10 | toddf | http://old.www.freeworlddialup.com/support/quick_start_guide for gandhijee |
| 22:20.36 | *** join/#asterisk |Vulture| (n=V@216.84.158.116) |
| 22:21.29 | websae2k | so the RTP ports that are set on the SIPURA need to be forwarded? |
| 22:23.00 | gandhijee | toddf: i didn't think that when they said dial 393612 that the 393 wasn't supposta be part of it |
| 22:23.10 | Ariel_ | Down here there is a new one. Called Thebluezone |
| 22:23.25 | Qwell | Ariel_: related to theblue? heh |
| 22:23.41 | Ariel_ | Qwell, no |
| 22:24.12 | gandhijee | toddf: and it seems only 2 of those numbers work, the time and echo test. |
| 22:24.35 | websae2k | the only way i can seem to get my phone to be operational is set the EXTERNIP in the SIPURA'S CONFIG...but my ip address is static |
| 22:25.13 | websae2k | any ideas...anyone? |
| 22:25.28 | websae2k | so once my ip address changes the phone won't work |
| 22:25.42 | Qwell | if its static, it won't change |
| 22:25.46 | websae2k | and i'll go back dto being able to call out but not hear anyone on the other end but them hear me |
| 22:25.52 | websae2k | it's dynamic* |
| 22:25.58 | websae2k | mistyped |
| 22:27.33 | websae2k | so i don't know...what to do |
| 22:27.37 | websae2k | i can also recieve calls |
| 22:27.44 | websae2k | i just never hear the person on the other end |
| 22:27.47 | websae2k | they can hear me |
| 22:28.49 | harryvv | web, its your firewall and or your sip/iax configuration now allowing sip/rtp to pass |
| 22:29.11 | harryvv | more then likly its your firewall |
| 22:29.13 | websae2k | so i can recieve calls though |
| 22:29.25 | harryvv | yes..thats typical |
| 22:29.42 | harryvv | but you wont hear them..your firewall is blocking there incomming voice. |
| 22:33.35 | *** join/#asterisk marc324 (n=marc3234@206-248-159-4.dsl.teksavvy.com) |
| 22:34.17 | morale | <PROTECTED> |
| 22:34.17 | morale | ls |
| 22:34.17 | morale | w |
| 22:35.13 | rayvd | I enjoy chatting. |
| 22:35.14 | rayvd | About grain. |
| 22:35.25 | websae2k | how do I know what the RTP ports are? |
| 22:36.13 | patpatnz | harryvv, iax doesn't require seperate rtp ports does it? |
| 22:36.25 | patpatnz | harryvv, isn't it all trunked along one connection? |
| 22:37.04 | *** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net) |
| 22:37.37 | websae2k | here's the thing the only way i can have it so I can hear people talking to me, and not just them hearing me...is to set the EXT IP on my SIP phone |
| 22:37.42 | professorchen | in relation to openpbx I am looking for an addon names res_data or ast_data, which some users say does a much better job than Asterisk Realtime, but doesn't appear in the add on list anymore, does anyone know where it is? |
| 22:38.31 | websae2k | so RTP can't be the issue...if i setup EXT IP on SIPURA phone and it works fine |
| 22:38.36 | websae2k | but i have a dynamic ip address |
| 22:38.39 | websae2k | so that's an issue |
| 22:38.43 | wunderkin | you're in the wrong channel.. wow already? i've heard of res_data but haven't seen it used for awhile |
| 22:39.37 | FuriousGeorge | isnt there some way to do an xfer with # at least on zap channels, and if so, how does that work with IVRs |
| 22:39.48 | FuriousGeorge | that you call |
| 22:40.30 | marc324 | what editor is everyone using for modifying the config files? |
| 22:40.45 | Mother_ | vi? |
| 22:40.51 | justinu | vi |
| 22:40.59 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 22:41.03 | Qwell | ed |
| 22:41.35 | wunderkin | pico! |
| 22:42.01 | marc324 | im new to * |
| 22:42.14 | wunderkin | to unix too? |
| 22:42.30 | marc324 | ~ |
| 22:42.30 | FuriousGeorge | marc324: use nano |
| 22:42.34 | FuriousGeorge | marc324: if ur new |
| 22:42.42 | wunderkin | not being rude but a text file is a text file, are you asking for a gui? |
| 22:42.42 | Mother_ | they usually go hand-by-hand |
| 22:42.48 | marc324 | command not found |
| 22:42.54 | FuriousGeorge | marc324: install nano |
| 22:43.01 | wunderkin | Mother_: not all of the time :P :P its just an editor |
| 22:43.07 | Mother_ | heheh |
| 22:43.18 | FuriousGeorge | marc324: switch to gentoo. it installs nano by default :) |
| 22:43.20 | websae2k | please.......anyone......have any idea...why i can call out but the person can hear me, i can't hear them.........the only way i can get it so i can hear them is to set the EXT IP field in my SIPURA 841 (hardphone) to my public ip address, but it's dynamic so that couuld be problematic...any ideas anyone...greatly appreciated :) |
| 22:43.28 | marc324 | is the new book on * worthit? |
| 22:43.31 | Mother_ | marc324: vi is rather simple to use....no need for anything more complex |
| 22:43.42 | FuriousGeorge | websae2k: www.dyndns.org |
| 22:43.42 | wunderkin | simple? |
| 22:43.59 | Mother_ | vi textfile, insert, type :wq to save and quit, or :w to save, :q! to quit without saving changes |
| 22:44.00 | *** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net) |
| 22:44.09 | tzanger | heh |
| 22:44.13 | wunderkin | nano is much easier, all you have to know is ^ is control |
| 22:44.18 | tzanger | I used to open another console and kill vi when I was first learning linux |
| 22:44.22 | tzanger | vi is so nice now |
| 22:44.28 | Mother_ | lol indeed |
| 22:44.29 | tzanger | I put gvim on my windows machines now |
| 22:44.35 | FuriousGeorge | wunderkin: you dont even need to know that to use it |
| 22:44.37 | tzanger | there is a learning curve but it's well worth it |
| 22:45.07 | RaYmAn-Bx | imho the problem with vi is that you need to remember loads of shortcuts to use it even vaguely effiently :P |
| 22:45.12 | wunderkin | FuriousGeorge: how do you know what to do after typing in text then? |
| 22:45.32 | wunderkin | RaYmAn-Bx yes |
| 22:45.39 | FuriousGeorge | oh yeah, you gotta close it i guess |
| 22:45.48 | Mother_ | just like anything else really |
| 22:46.07 | Mother_ | you need to know a ton of asterisk-related things to use it efficiently...everything has a learning curve |
| 22:46.14 | FuriousGeorge | but just like everything else its ctrl+x |
| 22:46.23 | FuriousGeorge | i mean c |
| 22:46.36 | wunderkin | control x is exit yes |
| 22:46.45 | FuriousGeorge | i guess its X. |
| 22:46.49 | wunderkin | control c shows cursor position |
| 22:46.58 | wunderkin | somewhat backwards yes |
| 22:47.23 | FuriousGeorge | pop quiz: true or false: if i want to make the outside party quieter on a zap channel i lower my rcgain to -1.0? |
| 22:47.38 | tzanger | that makes them 1dB quieter yes |
| 22:47.39 | FuriousGeorge | *rxgain (in zapata.conf) |
| 22:47.49 | FuriousGeorge | tzanger: yee haw |
| 22:49.08 | *** join/#asterisk zotz (n=zotz@24.231.36.100) |
| 22:50.37 | *** join/#asterisk blkbearnh (i=turner@c-24-147-155-3.hsd1.nh.comcast.net) |
| 22:50.57 | websae2k | i tried forwarding RTP ports 10000-20000 but still could not hear person |
| 22:51.00 | blkbearnh | Evening all, can someone help a writer on deadline? |
| 22:51.00 | websae2k | they could only hear me |
| 22:51.01 | sahafeez | I have a context called Internal |
| 22:51.05 | websae2k | i am trying to figure that out |
| 22:51.07 | websae2k | any ideas |
| 22:51.18 | websae2k | i forwarded those RTP ports...still ddidn't work |
| 22:51.23 | sahafeez | for extend to extend |
| 22:51.25 | sahafeez | [internal] |
| 22:51.25 | sahafeez | exten => _42XX,1,Dial,sip/${EXTEN}|30|to |
| 22:51.26 | sahafeez | exten => _42XX,2,Voicemail,u${EXTEN} |
| 22:51.31 | websae2k | only way was to set EXT IP in my sip phone settings |
| 22:51.31 | sahafeez | it never rings. |
| 22:51.34 | websae2k | which ishouldn't have to do |
| 22:51.35 | sahafeez | what did i miss |
| 22:51.41 | websae2k | i use to not have to do that with a server i was on |
| 22:52.06 | *** join/#asterisk paryl (n=paryl@209.236.78.59) |
| 22:52.09 | blkbearnh | Can someone help me with basic setup of a TDM31B? |
| 22:52.28 | websae2k | so RTP ports are forwarded...stil can't hear person, they can only hear me....only way to make it work is to set the EXT IP in my sipura hard phone's config...but i never had to do that before with the server i connected to |
| 22:52.36 | paryl | so i see "Started music on hold, class 'default', on SIP/4082-f977" in the console, but i hear nothing. |
| 22:52.39 | websae2k | anyone's help still appreciated :) |
| 22:55.07 | *** join/#asterisk SkramX (n=skramy@vistech.org) |
| 22:56.13 | sahafeez | ok. figured it out |
| 22:56.38 | wunderkin | blkbearnh: making a competing asterisk book? heh |
| 22:56.48 | blkbearnh | Nope, reviewing it for Linux Journal |
| 22:57.18 | blkbearnh | But I can't get the channels on the TDM31B to show from show channels |
| 22:57.58 | wunderkin | oh, did you follow the guide on digium.com? if you go to support, docs, somethin like that they should have setup instructions.. |
| 22:58.07 | blkbearnh | I did |
| 22:58.30 | blkbearnh | Oct 6 18:46:15 WARNING[26484]: chan_zap.c:9651 setup_zap: Ignoring switchtype |
| 22:58.30 | blkbearnh | <PROTECTED> |
| 22:58.30 | blkbearnh | <PROTECTED> |
| 22:58.30 | blkbearnh | <PROTECTED> |
| 22:58.31 | blkbearnh | <PROTECTED> |
| 22:58.31 | blkbearnh | <PROTECTED> |
| 22:58.33 | blkbearnh | <PROTECTED> |
| 22:58.35 | blkbearnh | <PROTECTED> |
| 22:58.37 | marc324 | is there a copy of the config files stored elsewhere? |
| 22:58.42 | blkbearnh | *CLI> show channels |
| 22:58.42 | blkbearnh | <PROTECTED> |
| 22:58.42 | blkbearnh | 0 active channel(s) |
| 22:58.56 | tzanger | blkbearnh: looks goo dso far |
| 22:58.56 | JunK-Y | blkbearnh: its normal theure not active. |
| 22:59.08 | blkbearnh | No dialtone on a phone attached to it |
| 22:59.08 | wunderkin | yeah that only shows active channels |
| 22:59.09 | JunK-Y | zap show channels |
| 22:59.29 | blkbearnh | *CLI> zap show channels |
| 22:59.29 | wunderkin | all it shows is you installed umm whats it called.. since tor shows up :) |
| 22:59.29 | blkbearnh | <PROTECTED> |
| 22:59.29 | blkbearnh | <PROTECTED> |
| 22:59.29 | blkbearnh | <PROTECTED> |
| 22:59.29 | blkbearnh | <PROTECTED> |
| 22:59.30 | blkbearnh | <PROTECTED> |
| 22:59.32 | blkbearnh | <PROTECTED> |
| 22:59.46 | JunK-Y | blkbearnh: bingo, now have fun:) |
| 22:59.49 | blkbearnh | But why no dialtone? |
| 23:00.01 | JunK-Y | blkbearnh: goto www.voip-info.org |
| 23:00.17 | blkbearnh | Been there |
| 23:00.29 | Qwell | ~pb |
| 23:00.31 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 23:00.55 | blkbearnh | Sorry, jbot :-) |
| 23:03.21 | sahafeez | [cdr_addon_mysql.so] => (MySQL CDR Backend) |
| 23:03.29 | tzanger | when you pick up a phone in port 1 2 or 3 do you see "starting simple switch on Zap/1" (for port 1, for example) ? |
| 23:03.35 | sahafeez | where is this refernced in the startup so I can comment it out |
| 23:03.43 | *** join/#asterisk micc (n=dotirc@c-24-19-175-112.hsd1.wa.comcast.net) |
| 23:03.55 | blkbearnh | Nope |
| 23:04.06 | marc324 | is there a copy of the config files... i read it was in the src dir? |
| 23:04.07 | tzanger | sahafeez: ahh, young grasshopper, that is where you just proved to us you haven't read anything. Go, read. |
| 23:04.20 | blkbearnh | WOuld I see that on the CLI? |
| 23:04.23 | tzanger | blkbearnh: are teh lights on the back of hte card green? |
| 23:04.25 | micc | I'll pay someone $100 to show me an iax connection from a windows client to asterisk that doesn't skip audio. |
| 23:04.26 | blkbearnh | Yep |
| 23:04.49 | tzanger | blkbearnh: and you have a phone plugged in to one of the first three ports (top of the PCI card is port #1) |
| 23:04.59 | blkbearnh | Does it matter that I don't have a POTS line plugged in yet? (yes, tzanger) |
| 23:05.07 | tzanger | blkbearnh: no, don't plug in the POTS line yet |
| 23:05.15 | sahafeez | tzanger: i have read to much and i am brain fried and the docs are the most un-organzied pile i have ever seen |
| 23:05.16 | micc | Anyone in the Seattle area have a working iax windows client? |
| 23:05.17 | tzanger | you have very basic issues, but I am not sure what it is yet :-) |
| 23:05.41 | blkbearnh | When I take the phone off the hook, I do hear a click, so there's power there |
| 23:05.44 | tzanger | sahafeez: well if you don't understand then and we just tell you what to do to get the default dialplan to go away you'll be here within 30 seconds asking how to change the next trivial thing. |
| 23:05.52 | tzanger | blkbearnh: yes but you didn't see anything on the CLI |
| 23:05.52 | marc324 | how do i copy all files in one directory to a subdirectory in that same directory? |
| 23:05.56 | blkbearnh | Nope |
| 23:06.05 | tzanger | pastebin your /etc/asterisk/zapata.conf and /etc/zaptel.conf |
| 23:06.05 | tzanger | please |
| 23:06.16 | sahafeez | i have everything up and work thanks. just this last error. |
| 23:06.34 | *** join/#asterisk implicit (n=implicit@dhcp-236-141.mobile.uci.edu) |
| 23:06.43 | blkbearnh | On pastebin? |
| 23:07.56 | blkbearnh | zapata.conf is up |
| 23:07.56 | *** join/#asterisk lancey (i=Shady@support.net1.cc) |
| 23:07.58 | lancey | hi guys |
| 23:08.05 | blkbearnh | http://pastebin.ca/24772 |
| 23:08.25 | lancey | i've got a problem setting CID on outgoing SIP calls, is there anything special about it? |
| 23:08.42 | marc324 | what is the pathname of a subdirectory |
| 23:09.25 | blkbearnh | http://pastebin.ca/24773 for /etc/zapata.conf |
| 23:10.19 | marc324 | how do you display the full pathname in the shell? |
| 23:10.53 | Qwell | pwd |
| 23:11.08 | micc | are all soft phones just crap? |
| 23:11.31 | micc | I should clarify. Are all windows soft phones crap? |
| 23:11.39 | lancey | micc, firefly sort of works |
| 23:12.03 | micc | lancey, firefly seems to be the worst. It has the same audio breakups as the rest of them, but it can't seem to keep a consistent pitch. |
| 23:12.23 | micc | or tone or speed. |
| 23:12.55 | lancey | ?! |
| 23:13.00 | lancey | never had any problems with it |
| 23:13.08 | lancey | the sound is just ok |
| 23:13.20 | micc | lancey, you've never had it drop audio on you? |
| 23:13.26 | lancey | nopes |
| 23:13.39 | micc | what provider do you use? |
| 23:13.40 | FuriousGeorge | marc324: ls --help |
| 23:13.44 | FuriousGeorge | cp --help |
| 23:13.47 | lancey | how does this matter? |
| 23:13.55 | lancey | i'm connecting it to an * box |
| 23:14.04 | lancey | and then calls go through different providers |
| 23:14.25 | lancey | i can say for sure it works fine at least with iLBC and alaw |
| 23:15.29 | marc324 | the extensions.conf is full... ok to delete all? |
| 23:17.46 | sahafeez | if i no via DID that the call is a fax, can i map it out without the Answer |
| 23:18.47 | sahafeez | s/no/know |
| 23:18.55 | hardwire | http://scoreboard.keynote.com/scoreboard/Main.aspx?Login=Y&Username=public&Password=public |
| 23:18.56 | hardwire | :( |
| 23:20.18 | marc324 | how can i find the current asterisk version? |
| 23:20.23 | sahafeez | cvs |
| 23:21.02 | lancey | marc324 it seems u need a check of http://www.voip-info.org and http://www.asterisk.org at first |
| 23:23.20 | marc324 | whats a zap? |
| 23:25.27 | Mother_ | it's a think for killing bugs like flies etc |
| 23:25.38 | Mother_ | s/think/thing |
| 23:26.16 | SkramX | marc324: physical digium card |
| 23:26.36 | JunK-Y | its a type of channel driver |
| 23:26.39 | JunK-Y | show channeltypes |
| 23:27.24 | *** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net) |
| 23:27.35 | JunK-Y | in ur CLI, type help and read a lot :) |
| 23:27.48 | lancey | :) |
| 23:27.59 | Mother_ | type help and look up each line that comes up in the wiki ;) |
| 23:28.00 | *** join/#asterisk ManxPower (i=eric@1Cust596.an1.dfw28.da.uu.net) |
| 23:28.12 | Mother_ | hi manx |
| 23:28.26 | ManxPower | Do any of you isp.monkeys have a temp dialup I can use to test outbound dialup from my cisco router? |
| 23:28.34 | lancey | anyone had any problems with setting CID of outgound SIP calls? |
| 23:28.52 | ManxPower | lancey, only if you use quotes |
| 23:28.53 | Mother_ | ManxPower: if you don't mind dialing international I can set one up |
| 23:29.04 | *** join/#asterisk acidfoo (n=acidfoo@66.11.160.156) |
| 23:29.08 | ManxPower | Mother_, I mind. |
| 23:29.11 | ManxPower | I'm in the USA |
| 23:29.12 | lancey | ManxPower if i use quotes it works, or if i use quotes it doesn't? |
| 23:29.14 | Mother_ | I thought so ;) |
| 23:29.21 | ManxPower | lancey, what specific CID problem do you have? |
| 23:29.34 | lancey | ManxPower i have a sip peer checking CID |
| 23:29.41 | *** join/#asterisk syle (n=blag@unaffiliated/syle) |
| 23:29.42 | ManxPower | lancey, no. Some sip devices don't work if they get CID in quotes |
| 23:29.52 | lancey | i've created an extension which firstly sets the correct CID |
| 23:29.57 | ManxPower | callerid=Robert Dobbs <666> |
| 23:30.00 | lancey | then dial from another * box |
| 23:30.01 | ManxPower | notice the lack of quotes |
| 23:30.18 | lancey | and the sip peer rejects my call, as the callerid is wrong |
| 23:30.31 | lancey | Verbose(${CALLERID}) shows the right one, though |
| 23:30.40 | ManxPower | Since nobody in the USA can help, I guess I'll have to sign up with ATT World Net |
| 23:31.02 | lancey | ManxPower i have a dial-in, but it won't work from USA, sorry :/ |
| 23:31.02 | ManxPower | lancey, what device is the SIP peer? |
| 23:31.14 | lancey | ManxPower dunno, probably a Cisco router |
| 23:31.32 | ManxPower | lancey, what is the callerid= line in your sip.conf? |
| 23:31.50 | lancey | there is no callerid= in sip.conf |
| 23:31.55 | lancey | sip is used for outbound only |
| 23:32.26 | lancey | calls get to me through iax, and in iax2.conf theres the right callerid line |
| 23:33.12 | lancey | i'm too much confused, as it works when i dial in my * box from an IAX2 phone |
| 23:33.31 | lancey | but dialing in from another * box with the same account gets rejected by the SIP peer |
| 23:35.14 | lancey | i'm now updating the other * box to latest CVS and will see what happens... |
| 23:35.26 | *** join/#asterisk menger (n=menger@dsl-53.69.240.220.rns01-dryb-mel.dsl.comindico.com.au) |
| 23:38.23 | *** join/#asterisk jeremywhiting (n=jeremy@71-37-101-103.slkc.qwest.net) |
| 23:40.11 | *** join/#asterisk SarahEmm (n=sarahemm@Toronto-HSE-ppp3682154.sympatico.ca) |
| 23:40.43 | *** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr) |
| 23:42.02 | *** join/#asterisk MnxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net) |
| 23:42.11 | MnxPower | at least my wireless internet came back |
| 23:47.26 | lancey | :) |
| 23:47.36 | lancey | MnxPower i believe http://bugs.digium.com/view.php?id=5325 has something to do with my problem :) |
| 23:48.12 | marc324 | can I delete all content of extensions.conf after installation? I only need to build a voicemail. |
| 23:48.18 | SarahEmm | err |
| 23:48.21 | SarahEmm | you need an extensions.conf |
| 23:48.26 | SarahEmm | you can delete the sample and make your own |
| 23:48.28 | SarahEmm | you still need one tho |
| 23:48.35 | lancey | marc324 -> www.voip-info.org PLEASE |
| 23:48.40 | FuriousGeorge | SarahEmm: i think he wants to ask if he should start from scratch |
| 23:48.44 | lancey | it's overexplained out there |
| 23:48.45 | marc324 | yes. |
| 23:48.50 | marc324 | scratch |
| 23:48.56 | FuriousGeorge | yes you should |
| 23:49.00 | FuriousGeorge | marc |
| 23:49.03 | FuriousGeorge | ~docs |
| 23:49.05 | 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 |
| 23:49.06 | lancey | marc324 the provided extensions.conf is just an example |
| 23:49.28 | Ariel_ | and the sample has a great macro for voicemails |
| 23:49.50 | *** join/#asterisk jskcr (n=jskcr@unaffiliated/jskcr) |
| 23:49.53 | *** join/#asterisk pc2 (n=pc@209.151.52.81) |
| 23:49.55 | pc2 | hmmm -- |
| 23:49.56 | pc2 | *CLI> Oct 6 19:53:25 NOTICE[2111]: chan_iax2.c:6629 socket_read: Rejected connect attempt from x.x.x.x, who was trying to reach '18663694977@ ?? |
| 23:49.58 | pc2 | What does that mean? |
| 23:50.53 | lancey | pc2 non-existing extension |
| 23:50.59 | lancey | probably... |
| 23:51.15 | lancey | or x.x.x.x missing in your iax2.conf |
| 23:53.14 | marc324 | how do you specify the number of rings before call is answered? |
| 23:53.48 | Icemaann | i have a context called home in exntensions.conf. I have a friend in sip.conf that can login correctly using x-lite. however, when dialing an extension in the home context it says address imcomplete (484 error). When I change context to default, the default extensions work fine. Thoughts? |
| 23:53.54 | Icemaann | using 1.2 beta btw |
| 23:54.03 | Mother_ | READ THE DOOOOOOOCS!!!!! |
| 23:54.09 | justinu | lol |
| 23:54.30 | justinu | read the docs or hire a consultant... |
| 23:54.30 | Mother_ | excuse me while I slit my wrists with a paperclip |
| 23:54.34 | justinu | lazy fuck |
| 23:55.30 | Icemaann | uh, was that aimed towards me? |
| 23:55.34 | hardwire | I have my last damn buck to the coffee lady |
| 23:55.36 | justinu | no |
| 23:55.40 | Ariel_ | Icemaann, you need to include the correct context |
| 23:55.42 | obsidian-studios | harsh in here tonight |
| 23:55.50 | hardwire | I can sorta feel the tension |
| 23:55.52 | justinu | he's been asking these questions all day |
| 23:56.02 | hardwire | I say we all unregister and try that other asterisk channels |
| 23:56.04 | Ariel_ | marc324, dial(Sip/100,20) 20 is about 4 rings |
| 23:56.10 | sleepy_one | consultant here at your service :-) |
| 23:56.17 | justinu | lol, see... i knew someone would step up |
| 23:56.25 | obsidian-studios | justinu: ah, sorry, but I have been there and done that, usually was ignored or beaten to death with reference to wiki :) |
| 23:56.30 | Icemaann | Ariel_: include it in default? or include default in home? the home context has all the extensions I need. its all in that context |
| 23:56.41 | hardwire | Ariel_: I had that question the other day.. I told somebody it times out after 20 seconds.. and they told me "well how many rings is that?" |
| 23:56.45 | hardwire | about 18 from the first ring :) |
| 23:56.48 | hardwire | err |
| 23:56.52 | hardwire | about 18 seconds :) |
| 23:57.17 | hardwire | justinu: who we picking on today? |
| 23:57.23 | justinu | marc |
| 23:57.31 | Ariel_ | hardwire, well 20 sec is about 4 rings in the us. But in other locations you never know |
| 23:57.38 | justinu | dude, we know you're a n00b, but come on... read a little bit |
| 23:57.41 | hardwire | Ariel_: sure.. over zap |
| 23:57.45 | Mother_ | it depends on ring cadence... |
| 23:57.51 | hardwire | my phones default to a wittle "beep" |
| 23:57.51 | Ariel_ | Icemaann, you should never have anything of value in the default. |
| 23:57.57 | hardwire | that happens quite fast |
| 23:58.07 | jskcr | anyone running asterisk on gentoo |
| 23:58.21 | SarahEmm | lots of us jskcr |
| 23:58.22 | SarahEmm | why? |
| 23:58.22 | Ariel_ | But setup your context and then include them as what your want the devices to do. Read the sample extensions.conf.sample located in the /usr/src/asterisk/configs |
| 23:58.24 | Icemaann | Ariel_: i dont, its all in home ;-) I think im getting somewhere now |
| 23:58.25 | hardwire | marc324: are you a noooob? |
| 23:58.31 | hardwire | my dog likes noooobs |
| 23:58.36 | Mother_ | I'd like to make something like app_random_cadence.c to run users crazy |
| 23:58.37 | obsidian-studios | jskcr: never :) |
| 23:58.37 | Icemaann | Ariel_: k ill dbl check |
| 23:58.52 | Mother_ | tuuu tuuuuuu tutut.....tuuut.....tut.... |
| 23:58.57 | jskcr | SarahEmm: I was wondering if anyone was running the 1.20 beta frin portage |
| 23:58.59 | Ariel_ | Icemaann, what have everything in home. |
| 23:59.03 | jskcr | err frin/from |
| 23:59.28 | *** join/#asterisk joat (n=joat@laketaylor.org) |
| 23:59.55 | SarahEmm | jskcr: i run CVS HEAD... |
