| 00:00.03 | bkw_ | i'm to the credit card screen.. bet a temp database issue |
| 00:00.53 | _chad | Asterisk 2.0 is the recent stable release? |
| 00:01.08 | Darwin[laptop] | 1.0.7 is |
| 00:01.21 | Darwin[laptop] | 2.0 is not even close to being done |
| 00:01.32 | Sedorox | bkw_: why do you need someone on Nextel with sms? |
| 00:01.39 | MrBelvedr | 2.0 is being rewritten in c# |
| 00:01.43 | MrBelvedr | :) |
| 00:01.53 | _chad | ah i just saw the note on the wiki about 2.0 stable being released or something |
| 00:01.55 | _chad | odd |
| 00:01.55 | Darwin[laptop] | what |
| 00:02.08 | PBXtech | chad: thats an april fools joke :) |
| 00:02.08 | MrBelvedr | that was an April 1 joke |
| 00:02.09 | Darwin[laptop] | c# |
| 00:02.14 | _chad | ah |
| 00:02.14 | Darwin[laptop] | ahh ok |
| 00:02.15 | _chad | lol |
| 00:02.16 | marlowe | lol |
| 00:02.19 | marlowe | Asterisk 3.0 is PHP |
| 00:02.28 | marlowe | And what was 2.5 ? Java? |
| 00:02.30 | Sedorox | and 2.5 is Java... |
| 00:02.32 | Sedorox | yes |
| 00:02.33 | Sedorox | hehe |
| 00:02.36 | Sedorox | lots of rewrites |
| 00:02.42 | _chad | 4.0 in basic |
| 00:02.42 | Darwin[laptop] | I thought they where rewriting it in visual basic |
| 00:02.46 | _chad | lol yah |
| 00:02.54 | MattH | oooooooo basic |
| 00:03.03 | Darwin[laptop] | or cobal |
| 00:03.25 | Sedorox | 10 Print "welcome to asterisk" |
| 00:03.31 | Sedorox | 20 goto 10 |
| 00:03.32 | Sedorox | :-p |
| 00:03.35 | che | hehe |
| 00:03.37 | che | run ;) |
| 00:03.39 | MrBelvedr | hehe |
| 00:03.47 | MrBelvedr | cobol.net |
| 00:03.50 | che | Sedorox, had a c64 too? ;) |
| 00:03.53 | MrBelvedr | by fujitsu corp |
| 00:04.04 | Sedorox | naa... did have a AppleII tho once... |
| 00:04.25 | Darwin[laptop] | hehhe |
| 00:05.49 | Darwin[laptop] | what about pascal |
| 00:06.01 | Shido6 | Load "*",8,1 |
| 00:06.07 | Shido6 | RUN |
| 00:06.14 | Shido6 | SYS98744 |
| 00:06.27 | marlowe | u know i reall gotta put my router on a UPS |
| 00:06.32 | che | Shido6, first id load"$",8 and list ... then id look for a speedloader ;) |
| 00:06.47 | marlowe | thats the 5th time my power went out |
| 00:06.50 | Shido6 | push the red button on the Mach5 |
| 00:07.19 | Shido6 | wait a minute, flip the disk on the 1541 drive |
| 00:07.28 | che | Shido6, i had a 1571 ;) |
| 00:07.39 | Shido6 | I had 2 1541s and 1 1571 |
| 00:07.54 | che | Shido6, the datasettes were evil ;) |
| 00:07.55 | Shido6 | the 1541 came with the c64 and bought another then had the 1571 with the c128 |
| 00:08.05 | Shido6 | 45 minutes load time on the casettes |
| 00:09.11 | che | but its impressive whats they did with 64kb ram ;) |
| 00:09.18 | Sedorox | hehe |
| 00:09.48 | MrBelvedr | fuck |
| 00:09.59 | MrBelvedr | none of the call limiting directives are working |
| 00:10.11 | MrBelvedr | S and L are both ignored |
| 00:10.50 | che | Shido6, i got a nice c64 stream url want it? |
| 00:11.03 | Shido6 | stream? |
| 00:11.13 | che | Shido6, just dont wanna throw it in chan cause i dont know how many users it takes. well sound stream ;) c64 sound |
| 00:11.22 | Shido6 | ok |
| 00:12.11 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 00:12.45 | Juxt | is there a way to simulate the "transfer" button by dialing some sort of a sequence? |
| 00:12.57 | marlowe | # ? |
| 00:13.12 | Juxt | yeah for some reason it doesn't work on firefly, weird |
| 00:13.30 | marlowe | something wrong with the user |
| 00:13.36 | marlowe | it works on firefly for me |
| 00:13.43 | rvhi | try to make * from 1.0.7 |
| 00:13.56 | rvhi | how do i NOT generate chan_modem.so |
| 00:13.57 | _chad | anyone know off hand what file the jitter buffer is defined in? |
| 00:13.58 | Juxt | so if i wanna transfer i dial # and then the extension to transfer the call to right? |
| 00:14.06 | rvhi | it causes problem when starting * |
| 00:14.09 | marlowe | #<exten># |
| 00:14.18 | marlowe | U dont need the last # but I use it.. its safer or some crap |
| 00:14.22 | marlowe | transfers faster |
| 00:14.24 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || Someone that uses sms on nextel,cricket please msg bkw_ |
| 00:14.26 | rvhi | can't find where to modify in channels/Makefile |
| 00:14.52 | Sedorox | bkw_: I have nextel... |
| 00:14.58 | marlowe | bkw_: I have nextel |
| 00:15.06 | marlowe | I also have eVerizon but I guess u dont need that anymore |
| 00:15.13 | bkw_ | ok |
| 00:15.20 | bkw_ | guess I can make nextel work next |
| 00:15.25 | Sedorox | lol |
| 00:15.26 | bkw_ | msg me your numbers |
| 00:15.30 | marlowe | I never herad of cricket |
| 00:15.36 | bkw_ | and i'll see |
| 00:15.36 | marlowe | lol |
| 00:15.42 | Sedorox | ditto |
| 00:15.45 | bkw_ | I have 5 providers working with app_websms.c now |
| 00:15.48 | Sedorox | must be aust. :-p |
| 00:15.53 | Sedorox | cool |
| 00:15.56 | marlowe | bkw_: Actually I left my nextel @ work - If u still need help on Monday I'll help you out |
| 00:16.04 | bkw_ | tmobile,sprint,verizon,alltel,attws |
| 00:16.21 | Sedorox | attws = cingular.... |
| 00:16.21 | marlowe | haha I got verizon,nextel, & tmobile |
| 00:16.22 | Sedorox | lol |
| 00:16.29 | Sedorox | why three? |
| 00:16.37 | marlowe | Verizon - General Cell Pone - Best Service |
| 00:16.42 | marlowe | Nextel - PTT - I hate the service here |
| 00:16.43 | _chad | ahh rockin, it looks like my jitter buffer is completely switched off.. wonder if that might equate to some of the trouble :) |
| 00:16.45 | marlowe | Tmobile - Sidekick |
| 00:16.46 | bkw_ | Sedorox, ya but i got it working with attws's mmode.com thingy |
| 00:16.54 | Sedorox | ah cool |
| 00:17.23 | Juxt | hmm does the record for the phone in iax.conf need to contain transfer? |
| 00:17.35 | Sedorox | I had BoostMobile and Cingular for a while.. till my cingular ran up.. I switched to nextel |
| 00:18.04 | story | cingular has bogus billing |
| 00:18.06 | story | they cheat u |
| 00:18.11 | Sedorox | they didn't me... |
| 00:18.40 | marlowe | Cingular SUCKS |
| 00:18.43 | _chad | at&t/cingular has horrid billing policies |
| 00:18.48 | Sedorox | hehe.. tmobile sucks |
| 00:18.54 | marlowe | tmobile does suck |
| 00:18.59 | marlowe | Im going to throw my sidekick against a wall soon |
| 00:19.35 | Sedorox | lol |
| 00:19.51 | marlowe | Im going to just get a blackberry or treo or something and use it on verizon |
| 00:19.58 | Sedorox | treo |
| 00:20.01 | story | verizon broadband wireless |
| 00:20.10 | marlowe | I have verizon broadband wireless too |
| 00:20.34 | Darwin[laptop] | how much is it a month |
| 00:20.42 | marlowe | 79.99 but i get a discount |
| 00:20.44 | Darwin[laptop] | and how well does it work |
| 00:20.45 | marlowe | I pay $50/mo |
| 00:20.46 | shodan | if I run 3x phones lines with one cat5 , am I going to get lots of crosstalk (using the right torsaded wire pair for each line) ? |
| 00:20.48 | marlowe | It's awesome |
| 00:20.51 | marlowe | I get 2megs where I live |
| 00:20.53 | story | how'd u get a discount |
| 00:20.55 | Darwin[laptop] | 79 a month |
| 00:21.00 | marlowe | story: I cant say |
| 00:21.00 | Darwin[laptop] | screw that |
| 00:21.04 | story | oh |
| 00:21.05 | story | sekret |
| 00:21.22 | marlowe | But go on www.howardforums.com |
| 00:21.22 | marlowe | You'll find a way |
| 00:21.32 | story | hehe ok ill check it out |
| 00:21.44 | _chad | $50/mo for unlimited bw marlowe? |
| 00:21.58 | tzanger | shodan: nah |
| 00:22.13 | tzanger | you might get some crosstalk when one line rings |
| 00:22.31 | marlowe | yeah chad |
| 00:22.33 | Majestik | Does anyone know if there is a good CDR reporter for asterisk using csv files? |
| 00:22.43 | _chad | marlowe thats pretty bangin, include voice time also? |
| 00:22.49 | marlowe | no |
| 00:23.01 | marlowe | unlimited internet, unlimited txt messaging |
| 00:23.07 | slePP | Majestik: use a database :> |
| 00:23.15 | shodan | k , that'll do a much cleaner job , I'll get the 3 phone lines straight to my asterisk box , then feed it to the rest of the house (until I find a way to have FXS ports at 25$/each)) |
| 00:23.24 | Juxt | what does this allmean Attempting native bridge of IAX2/richmedium8001@richmedium8001/1 and IAX2/richmedium8003/2 |
| 00:23.24 | Juxt | <PROTECTED> |
| 00:23.24 | Juxt | <PROTECTED> |
| 00:23.33 | _chad | marlowe, what kind of handset are you on? |
| 00:23.39 | Majestik | slePP: yeah, there is that.. but the deb package I'm running doesn't seem to make mysql logging as an option. |
| 00:23.45 | marlowe | I have a pcmcia card |
| 00:23.49 | marlowe | I used to use my motorola v710 via bluetooth to do it |
| 00:23.51 | marlowe | that worked as well |
| 00:23.56 | slePP | ah |
| 00:24.26 | Majestik | unless of course I don't know what I'm looking for to enable it.. |
| 00:24.26 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 00:24.32 | shodan | are there 100$ wifi voip phones yet ? |
| 00:24.50 | marlowe | shodan: On April 1st they were selling some |
| 00:24.59 | _chad | marlowe, sweet :) |
| 00:25.05 | marlowe | i personally prefer getting an IAXY + Cordless phone |
| 00:25.06 | marlowe | ;) |
| 00:26.21 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 00:27.06 | rvhi | i don't want to compile iax/iax2, how do i modify channels/Makefile to do it? |
| 00:27.42 | marlowe | Why dont you want to compile it? |
| 00:27.46 | marlowe | You hav something against it? |
| 00:28.22 | MrBelvedr | how can i "see" what strings a softphone is sending through the manager api? |
| 00:28.37 | file[laptop] | IAX2 is the worst protocol ever created! |
| 00:28.46 | MrBelvedr | is there wa way to sniff it through the CLI ? |
| 00:29.57 | MrBelvedr | ? |
| 00:30.22 | mishehu | oops! |
| 00:30.57 | rvhi | i just don't use it, so no need to load it, nothing against it |
| 00:31.21 | mishehu | rvhi: noload chan_iax or something like t hat |
| 00:31.23 | rvhi | also other modules, e.g. chan_modem_i4l.so |
| 00:31.45 | rvhi | i have to manual disable them in modules.conf |
| 00:31.52 | rvhi | how can i stop even compile |
| 00:32.11 | rvhi | the issue is that with new version and new modules added, i have to change modules.conf manually |
| 00:32.53 | marlowe | <file[laptop]> IAX2 is the worst protocol ever created! |
| 00:32.59 | marks__ | ANYONE HAVE A GOOD VOICE to do recordings for a pbx? PrivMsg Me.. |
| 00:33.09 | marlowe | marks__: Wtf do u need? |
| 00:33.18 | *** join/#asterisk WhiteWlf (WhiteWolf@CPE-69-76-133-249.kc.res.rr.com) |
| 00:33.19 | marks__ | Someone to record voice prompts |
| 00:33.20 | marlowe | Professionally? Youll pay |
| 00:33.34 | marlowe | 609-252-1155 listen to my phone system - if you like it - let me know what you want |
| 00:33.40 | marlowe | ill give y ou a quote |
| 00:34.04 | marlowe | i take it thats u |
| 00:34.05 | marlowe | lol |
| 00:34.06 | WhiteWlf | I'm having some troubles, I just installed and compiled asterisk and whenever it trys to play an audio file... it hangs (1.0.7) - does the soundcard need to be installed in order for playing to work? |
| 00:34.16 | file[laptop] | marlowe: actually it was me |
| 00:34.28 | marlowe | file[laptop]: u should listn to my MOH |
| 00:34.30 | marlowe | I just finished that |
| 00:34.40 | marlowe | It was a lot of work |
| 00:34.41 | file[laptop] | nah |
| 00:34.49 | marlowe | ok someones calling from a landline |
| 00:34.51 | mishehu | WhiteWlf: a soundcard is not required for using asterisk. |
| 00:34.51 | marlowe | I got cid info |
| 00:34.58 | marks__ | marlowe- not for pay.. unless uull take like 9 dollars |
| 00:34.58 | marlowe | barry? :) |
| 00:35.21 | PBXtech | .. per month for 24 months |
| 00:35.43 | *** part/#asterisk Juxt (user@sfl-dsl-64-135-113-4-cust.host.net) |
| 00:35.45 | marlowe | theres like 6 calls in lol |
| 00:35.47 | file[laptop] | eyebeam is not liking my laptop |
| 00:35.50 | marlowe | everyone in the channel is calling |
| 00:36.00 | file[laptop] | oh I know why |
| 00:36.12 | Sedorox | lol' |
| 00:36.21 | file[laptop] | silly me, I was using it with our media stuff and it supports silence suppression... asterisk doesn't, stupid me |
| 00:36.31 | WhiteWlf | mishehu: I was sure it wasn't but for example - when you call the VM system and it plays the happy little sound vm-theperson, it never proceeds past that point. |
| 00:36.34 | marlowe | thats right |
| 00:36.36 | marlowe | thats why xten is broke |
| 00:36.54 | WhiteWlf | mishehu: It won't even play that sound, even |
| 00:37.06 | mishehu | WhiteWlf: sounds like a different kind of problem. are you running * as root? |
| 00:37.12 | WhiteWlf | mishehu: I am |
| 00:37.15 | file[laptop] | there we go... |
| 00:37.53 | marlowe | my nme is matt |
| 00:37.57 | marlowe | if u wanna look me up in the directory |
| 00:38.00 | MrBelvedr | how can i see the commands that the softphone is sending to the manager api? |
| 00:38.03 | mishehu | WhiteWlf: don't know what the problem is |
| 00:38.12 | marlowe | it goes by first nme, not last name |
| 00:38.12 | WhiteWlf | mishehu: Nor do I... heh |
| 00:38.20 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || Someone that uses sms on cellone,cricket please msg bkw_ |
| 00:38.22 | mishehu | WhiteWlf: did you check the logs? |
| 00:38.23 | file[laptop] | there you are |
| 00:38.30 | marlowe | i dunno if i can answer |
| 00:38.32 | file[laptop] | too bad I have no mic |
| 00:38.34 | marlowe | my x-ten might be broke |
| 00:38.43 | WhiteWlf | mishehu: Going to... it's funny though... my other asterisk server works just fine (same hardware) |
| 00:38.43 | file[laptop] | hello? |
| 00:38.45 | WhiteWlf | let me look |
| 00:38.46 | file[laptop] | I hear you |
| 00:38.48 | file[laptop] | I wanted to hear the MOH |
| 00:38.50 | marlowe | one sec |
| 00:38.59 | file[laptop] | ah ads |
| 00:39.09 | marlowe | what else? :) |
| 00:39.09 | file[laptop] | appealing to the senses of your visitors! |
| 00:39.13 | marlowe | hahah |
| 00:39.15 | file[laptop] | ooh baby... you better believe I'm appealing |
| 00:39.25 | file[laptop] | back to music |
| 00:39.37 | marlowe | its 10 ads so far |
| 00:39.45 | marlowe | music is only played for 2 seconds |
| 00:39.52 | marlowe | each add is between 7 - 20 seconds |
| 00:39.53 | file[laptop] | blasphemy |
| 00:39.55 | marlowe | err, ad |
| 00:40.14 | shodan | is there a nice lightweight open source voip client that works with asterisk ? |
| 00:40.23 | marlowe | file[laptop]: did i sound clear? |
| 00:40.33 | file[laptop] | marlowe: a little background noise |
| 00:40.35 | marlowe | Just the other day i found out my laptop had an integrated microphone |
| 00:40.37 | file[laptop] | but I could hear you fine |
| 00:40.38 | luke-jr_ | shodan: Kphone |
| 00:40.44 | mishehu | shodan: you need to specify OS |
| 00:40.47 | marks__ | So.. anyone want to do the recordings for me? MESSAGE ME |
| 00:40.57 | luke-jr_ | mishehu: not really; all open source OS are fairly compatible |
| 00:41.00 | marlowe | drooth: You can talk in the channel |
| 00:41.03 | marlowe | 609-252-1155 |
| 00:41.06 | drooth | ok |
| 00:41.07 | drooth | thanks |
| 00:41.13 | *** join/#asterisk DannyF (~wizard@c-f8f472d5.020-103-73746f40.cust.bredbandsbolaget.se) |
| 00:41.24 | shodan | oops , I meant for windows , (I suppose there's a billion of them for linux) |
| 00:41.46 | file[laptop] | actually no. |
| 00:41.50 | luke-jr_ | shodan: what's the point? you're running a proprietary/immoral OS anyway... |
| 00:42.12 | tzanger | immoral |
| 00:42.12 | tzanger | hahaha |
| 00:42.41 | tzanger | I run PopeOS 2003 Papal Candacy 28 |
| 00:42.43 | shodan | I can't change my user , until I get a card with a couple fxs port I need voip software that run on my users' os |
| 00:42.44 | DannyF | what the heck is wrong with the cvs? |
| 00:42.51 | mishehu | luke-jr_: uhm, linphone does not work under win32, last I heard. |
| 00:43.05 | luke-jr_ | mishehu: Do you have a point? |
| 00:43.10 | DannyF | they redoing something? |
| 00:43.33 | mishehu | luke-jr_: the question is, do you have a point by this comment --> luke-jr_> mishehu: not really; all open source OS are fairly compatible |
| 00:43.46 | mishehu | you assume he was using an open source os. |
| 00:44.03 | luke-jr_ | mishehu: Sure; there's little purpose to caring whether a program is open source unless your OS is also |
| 00:44.06 | mishehu | shodan: donno whats opensource that works for win32... sorry. |
| 00:44.47 | mishehu | luke-jr_: uhm. I fail to see the logic in that. |
| 00:45.17 | mishehu | and you just prove the ass-u-me concept of assume. |
| 00:45.24 | *** join/#asterisk Dr-Linux (~sshah@202.163.69.3) |
| 00:45.56 | DannyF | *sigh* |
| 00:46.11 | Dr-Linux | does Asterisk support dialogic hardwares ? |
| 00:46.13 | mishehu | tzanger: isn't PopeOS up for a new major version release soon? |
| 00:46.54 | mishehu | Dr-Linux: not an expert on all hardware that works with asterisk, but there should be a list of what works with it on voip-info.org's wiki. |
| 00:46.54 | tzanger | yes, I think in 22 days |
| 00:46.55 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || Someone that uses sms on cellone,cricket or any other provider that has a web interface to send sms with please msg bkw_ |
| 00:47.06 | shodan | if I remove the opensource requirement , what's left in term of lightweight voip client that work with asterisk under windows ? |
| 00:47.09 | Qwell | tzanger: http://qdb.us/43460 |
| 00:47.34 | mishehu | shodan: you know, I think there might have been a build of iaxcomm or something that ran on win32. donno if it's still out there. |
| 00:47.51 | mishehu | that's open source I believe. |
| 00:48.21 | shodan | k , I'll try to find it |
| 00:48.38 | mishehu | the vatican should elect a jewish pope, that way we can put an end to the "is the pope jewish?" question-answering-a-question routine some people use frequently. |
| 00:48.49 | file[laptop] | bkw_: make telus work. |
| 00:48.55 | bkw_ | file[laptop], url? |
| 00:48.58 | bkw_ | and do you have a phone? |
| 00:49.12 | file[laptop] | yes I have a phone |
| 00:49.17 | file[laptop] | http://www.telusmobility.com/sendamessage/sendamessage.shtml |
| 00:49.18 | marlowe | fedex sucks |
| 00:50.10 | bkw_ | OMG those guys suck file |
| 00:50.16 | bkw_ | why do they try to split the damn npa nxx |
| 00:50.27 | bkw_ | thats just like rogers.. gonna make me redo the total framework here |
| 00:50.48 | Sedorox | hmmm |
| 00:51.05 | mishehu | marlowe: I usually have better luck with fedex than ups |
| 00:51.15 | marlowe | Fedex always screws me |
| 00:51.17 | marlowe | UPS never does |
| 00:51.27 | drooth | it's choppy a bit.. |
| 00:51.34 | marlowe | Whats coppy? |
| 00:51.40 | drooth | your out going |
| 00:51.40 | marlowe | choppy |
| 00:51.44 | marlowe | Ahh who cares |
| 00:51.46 | drooth | it breaks up a little |
| 00:51.51 | drooth | i care :) |
| 00:51.54 | marlowe | Im doing a lot of shit right now |
| 00:51.54 | marlowe | umm |
| 00:51.55 | drooth | QoS is important |
| 00:51.57 | file[laptop] | Fedex never screws me, UPS always does |
| 00:52.00 | drooth | ok |
| 00:52.01 | marlowe | Im not selling you service |
| 00:52.02 | file[laptop] | and not in the good way |
| 00:52.07 | yaboo | are all voip ports udp? |
| 00:52.08 | marlowe | So does it matter? |
| 00:52.19 | drooth | no but if you want to give people good impression.. |
| 00:52.29 | marlowe | They're not normally choppy.... |
| 00:52.36 | marlowe | Im doing, as I said, a lot of work on that server right now |
| 00:52.40 | drooth | ok |
| 00:53.00 | drooth | not bad |
| 00:53.01 | marlowe | wtf who hung up on me |
| 00:53.05 | marlowe | bitch |
| 00:53.06 | bkw_ | file did you get my sms? |
| 00:53.13 | file[laptop] | lemme go look |
| 00:53.18 | file[laptop] | one moment paleez |
| 00:53.31 | file[laptop] | yes |
| 00:53.43 | bkw_ | ok reworking my framework |
| 00:53.54 | bkw_ | blitzrage, rogers will follow |
| 00:55.51 | bkw_ | i'm about to just write a damn CGI and have two args passed.. provider and number |
| 00:55.52 | bkw_ | haha |
| 00:56.00 | file[laptop] | now now bkw |
| 00:56.02 | file[laptop] | :p |
| 00:56.30 | bkw_ | i'm gonna try something |
| 00:56.33 | bkw_ | I suspect one thing |
| 00:57.37 | bkw_ | file tell me if you got that one |
| 00:58.11 | file[laptop] | nothing yet |
| 00:58.20 | bkw_ | bet that didn't work |
| 00:58.25 | file[laptop] | nada |
| 00:58.31 | marlowe | This for nextel? |
| 00:58.36 | bkw_ | nextel works already |
| 00:58.38 | marlowe | oh |
| 00:58.39 | bkw_ | we are on telus now |
| 00:58.50 | bkw_ | i'm starting to think I just need a cgi |
| 00:58.55 | bkw_ | then link in the asterisk app to that |
| 00:58.59 | marlowe | nextel was fast, wow. :) |
| 00:59.07 | bkw_ | 6 providers work so far |
| 00:59.28 | bkw_ | i think i'm gonna just do a CGI |
| 00:59.29 | Moc | holly shit.. my app made my kernel segfault !!! |
| 00:59.37 | file[laptop] | silly Moc |
| 00:59.37 | bkw_ | kernel's don't segfault |
| 00:59.39 | bkw_ | they panic |
| 00:59.43 | bkw_ | haha |
| 00:59.45 | Moc | yes |
| 00:59.53 | bkw_ | moc working with meetme? |
| 00:59.59 | Moc | hold on.. |
| 01:00.02 | bkw_ | bad ioctl eh? |
| 01:00.15 | file[laptop] | lol I just glanced at the TV and thought it said "IAX" |
| 01:00.20 | bkw_ | haha |
| 01:00.39 | Moc | yes |
| 01:00.44 | Moc | donno yet |
| 01:01.25 | file[laptop] | Moc touched meetme in a naughty fashion |
| 01:01.33 | bkw_ | file i'm gonna regroup and convert all this to use a server side cgi to process the input from asterisk so it all will fall into the same framework |
| 01:01.33 | Sedorox | TV = $199.95, Cable = $50/month, Working too much on a FOSS project and thinking the TV mentioned soemthing from that project = Priceless |
| 01:01.36 | bkw_ | that would be the best way |
| 01:01.42 | yaboo | no matter either with iax or sip fwd numbers always are busy or congested from me when dialing them |
| 01:01.48 | file[laptop] | bkw_: that's cheap |
| 01:01.51 | file[laptop] | :p |
| 01:01.53 | Moc | question, does my box reboot or not |
| 01:01.57 | bkw_ | file[laptop], you wanna help me then eh? |
| 01:02.04 | bkw_ | file call 996 |
| 01:02.11 | file[laptop] | ah no no no :p |
| 01:02.17 | file[laptop] | you started this adventure, you finish it |
| 01:02.23 | file[laptop] | I'm watching Mythbusters anyway |
| 01:04.17 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 01:04.36 | bkw_ | i'll add npanxxparts |
| 01:14.22 | marlowe | wats 996 do ? |
| 01:15.16 | file[laptop] | it's where great minds gather to talk about evil plans of evilness |
| 01:15.30 | marlowe | awesome |
| 01:18.45 | WhiteWlf | how stupid |
| 01:22.13 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 01:25.39 | *** join/#asterisk Talmage (~Talmage@65.103.222.4) |
| 01:26.29 | Talmage | The release announcement on voip-info.org says asterisk 2.0 was released....for winblows....however, i notice the release date of april 1...was this an april fools joke or is asterisk really going to winblows? |
| 01:26.52 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
| 01:26.55 | jontow | Talmage; really.. do you need to ask that? :) |
| 01:27.13 | Talmage | ....i do need to ask....the boss wants to know |
| 01:27.17 | Talmage | and he has very little humor |
| 01:27.22 | jontow | *sigh* :) |
| 01:27.37 | jontow | no, no asterisk is not 'going to windows' |
| 01:27.51 | Talmage | has it been ported to c#? |
| 01:27.56 | PBXtech | Laughing Out Loud |
| 01:28.14 | PBXtech | you mean you belived 2 April fools jokes. hmmm |
| 01:28.16 | jontow | (alright.. i gave my portion of the answer.. i totally have to stop there) |
| 01:28.20 | Sedorox | Talmage: well... thats Asterisk 3.0 |
| 01:28.25 | Sedorox | and Asterisk 2.5 will be Java... |
| 01:28.26 | Talmage | great |
| 01:28.28 | Talmage | thanks... |
| 01:28.37 | file[laptop] | asterisk 3.5 will be written in basic |
| 01:28.38 | Sedorox | Note: I'm continuing April Fools |
| 01:28.43 | Sedorox | oh yes.. that too |
| 01:29.17 | jontow | i've been heavily pushing for one of the 3.x release-branches to be rewritten in complete Bourne (/bin/sh) shell script |
| 01:29.25 | jontow | that would effectively make my year. |
| 01:29.31 | file[laptop] | I bet |
| 01:29.44 | Talmage | great |
| 01:29.45 | Talmage | thanks |
| 01:29.46 | Talmage | .... |
| 01:29.56 | Talmage | I will now got let the air out of the boss's tires |
| 01:30.09 | file[laptop] | excellent idea |
| 01:30.13 | jontow | and place someone in a very high profile insane asylum |
| 01:30.15 | jontow | hah |
| 01:30.46 | Darwin[laptop] | ponders |
| 01:30.53 | Darwin[laptop] | brain on overload |
| 01:31.53 | file[laptop] | yay overload |
| 01:31.55 | Talmage | anyway....will be back someday....where you can laugh at me more |
| 01:31.59 | Talmage | thanks... |
| 01:32.25 | Talmage | one question before i go: how many before me have asked |
| 01:37.03 | Darwin[laptop] | does asterisk have a accujack protocal |
| 01:40.32 | Chuji | ~accujack |
| 01:41.43 | Darwin[laptop] | its a joke... |
| 01:42.12 | Darwin[laptop] | if you dont know what a accujack is go to a xxx store |
| 01:43.24 | Sedorox | Darwin[laptop]: I think we need more of a fufme protocol first |
| 01:44.21 | *** join/#asterisk Legend (~Legend@24.244.142.133) |
| 01:44.39 | *** join/#asterisk asteriskDOTbz (~logger@telux.net) |
| 01:44.40 | asteriskDOTbz | <PROTECTED> |
| 01:44.52 | marlowe | astlog: Fuck off |
| 01:44.53 | *** join/#asterisk HoopyCat (user@nocrtucker.netaccnt.net) |
| 01:45.14 | MrBelvedr | what is the name of the softphone with the slick looking UI? |
| 01:45.19 | HoopyCat | greetings |
| 01:45.21 | marlowe | jesus christ |
| 01:45.22 | MrBelvedr | the free one |
| 01:45.33 | Darwin[laptop] | xten |
| 01:45.39 | MrBelvedr | thx |
| 01:45.43 | HoopyCat | firefly's neater |
| 01:46.30 | MrBelvedr | do they both work with IAX? |
| 01:46.40 | HoopyCat | xten's SIP only; firefly works with IAX and SIP |
| 01:46.45 | MrBelvedr | I am looking for one that will show me the strings that it sends the Manager API |
| 01:47.31 | HoopyCat | hmmmm... that might be something else, as neither know anything about the manager api |
| 01:47.58 | MrBelvedr | how do they communicate with asterisk if not thru the manager api? |
| 01:48.10 | HoopyCat | using IAX :-) |
| 01:48.18 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 01:48.20 | MrBelvedr | so you can use IAX with sockets directly? |
| 01:48.34 | MrBelvedr | i don't get it |
| 01:48.44 | MrBelvedr | do they make connections direclty to the voip provider? |
| 01:48.52 | HoopyCat | it's UDP-based, so it will just... work, and stuff. |
| 01:48.57 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 01:49.14 | file[laptop] | MrBelvedr: I don't get what you're asking about |
| 01:50.50 | MrBelvedr | i have been using DIAX as my softphone. It connects to my local * box (which connects to Teliax) |
| 01:50.58 | file[laptop] | indeed. |
| 01:51.13 | MrBelvedr | so, I assumed that firefly and xten will also connect to my local * box (which connects to Teliax) |
| 01:51.19 | file[laptop] | yup. |
| 01:51.46 | MrBelvedr | my end goal it so somehow see the strings that the softphones are sending to my local * box |
| 01:51.59 | file[laptop] | iax2 debug will show you the IAX messages for firefly and DIAX |
| 01:52.03 | file[laptop] | for Xten you can do sip debug |
| 01:52.28 | MrBelvedr | I can't seem to find a way to see the strings being sent across the wire. I juat want to see the manager API commands, not all the low level IAX protocol stuff |
| 01:53.18 | file[laptop] | it's not using the manager API commands... |
| 01:53.21 | MrBelvedr | i just want to see the manager api commands |
| 01:53.23 | file[laptop] | manager doesn't come into it at all... |
| 01:53.50 | MrBelvedr | how do the softphones connect to my local * box then? |
| 01:53.57 | file[laptop] | using IAX or SIP |
| 01:54.08 | MrBelvedr | hmm |
| 01:54.16 | MrBelvedr | just uing raw sockets? |
| 01:54.23 | file[laptop] | it's just UDP packets |
| 01:54.45 | MrBelvedr | so what is the point of the manager api? what is the point? |
| 01:55.08 | file[laptop] | the manager API allows an outside application to interact with asterisk, see extension status, transfer calls, etc |
| 01:55.34 | MrBelvedr | right, DIAX and the softphones are outside applications, so why don't they use it |
| 01:55.46 | MrBelvedr | wait i am getting it |
| 01:55.52 | MrBelvedr | it is starting to make some sense |
| 01:56.04 | file[laptop] | IAX and SIP are for placing/receiving calls... and other stuff |
| 01:56.14 | file[laptop] | manager allows you to manipulate asterisk into doing things... |
| 01:56.57 | MrBelvedr | i see now |
| 01:57.11 | TomL | e.g. dynamically altering the dialing plan |
| 01:57.47 | file[laptop] | if you so want, sure |
| 01:57.53 | MrBelvedr | the only problem i am having now is that in my extension i have a DIAL command. But the S(1000)L(1000) directives are not being followed. THey should be limiting the call to 1 second but they are not! |
| 01:58.37 | MrBelvedr | it does not matter if i use a softphone or the manager api. the S and L directives are never limiting the call |
| 01:58.46 | MrBelvedr | anybody have a clue why |
| 01:59.07 | TomL | sorry, I never had a use yet for those features |
| 02:00.20 | MrBelvedr | has anybody done that |
| 02:01.20 | MrBelvedr | i am pulling hairs here |
| 02:01.41 | MrBelvedr | i only have one pube hair left |
| 02:02.25 | TomL | too much information |
| 02:02.50 | MrBelvedr | none left after that comment, thx Toml |
| 02:05.42 | MrBelvedr | can somebody try this on their setup and see if it works? exten => _1XXXXXXXXXX,1,DIAL(IAX2/thomasamiller@teliax/${EXTEN},20000,L(1000)) |
| 02:06.02 | *** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc) |
| 02:06.19 | MrBelvedr | just put the "L" directive on the end and see if your calls end after 1 second |
| 02:07.16 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 02:07.22 | ManxPower | ~docs |
| 02:07.23 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 02:08.32 | TomL | MrBelvedr: I don't think that's the right syntax |
| 02:08.45 | MrBelvedr | i have looked at all the docs, that is the right syntax |
| 02:08.55 | tzanger | haaaaaaaaaaahahahahahaha |
| 02:08.57 | tzanger | "On a traffic light green means go and yellow means yield, but on a banana it's just the opposite. Green means hold on, yellow means go ahead, and red means where the fuck did you get that banana at ..." |
| 02:09.20 | blitzrage | MrBelvedr: isn't that in seconds? |
| 02:09.33 | MrBelvedr | no millisecs |
| 02:09.43 | TomL | MrBelvedr: no, that is the wrong syntax for Dial() |
| 02:09.45 | MrBelvedr | i have tried it as L(1) |
| 02:09.50 | MrBelvedr | and L(1000) |
| 02:09.59 | TomL | exten => _1XXXXXXXXXX,1,DIAL(IAX2/thomasamiller@teliax/${EXTEN}|20000|L(1000)) |
| 02:10.00 | MrBelvedr | i have tried every possible combination |
| 02:10.10 | MrBelvedr | ok let me try TomL |
| 02:10.11 | TomL | you're using commas, it should be pipes |
| 02:10.13 | MrBelvedr | thanks |
| 02:10.14 | MrBelvedr | k |
| 02:10.15 | TomL | | not , |
| 02:10.17 | blitzrage | TomL: doesn't matter |
| 02:10.19 | Qwell | both work |
| 02:10.26 | TomL | hmm |
| 02:10.30 | TomL | k |
| 02:10.34 | blitzrage | TomL: | is actually the old syntax, but asterisk parces both equally |
| 02:10.39 | MrBelvedr | any other ideas? |
| 02:10.44 | blitzrage | MrBelvedr: could be a bug |
| 02:10.48 | blitzrage | let me try it |
| 02:10.51 | MrBelvedr | can sombody put the L(1000) and see if it works on yoursetyup |
| 02:10.53 | MrBelvedr | thank a million |
| 02:11.22 | blitzrage | MrBelvedr: well, I want to verify if its a bug |
| 02:11.29 | MrBelvedr | k |
| 02:11.53 | TomL | heh "old" syntax |
| 02:11.57 | TomL | CVS-HEAD-12/21/04-19:04:45 |
| 02:11.58 | TomL | :P |
| 02:12.47 | blitzrage | TomL: is that what you're running? |
| 02:12.50 | yaboo | followed the fwd.pulver site on setting up fwd with iax, but always get busy/congested when dialing a number other than the test 6xx numbers |
| 02:12.52 | TomL | yea |
| 02:12.55 | yaboo | any reason why |
| 02:13.01 | TomL | it never crashes, so I don't fuck with it :) |
| 02:13.07 | blitzrage | TomL: :D |
| 02:14.22 | *** join/#asterisk matiasg (~listas_as@200.68.82.225) |
| 02:14.26 | TomL | damn |
| 02:14.34 | matiasg | hi all |
| 02:14.44 | MrBelvedr | blitz did it work for you? |
| 02:14.53 | matiasg | has anyone got time for a weird question? |
| 02:15.27 | marlowe | MatsK: dont ask to ask |
| 02:15.29 | marlowe | just ask |
| 02:15.29 | matiasg | (regarding softphones and choppy sound..) |
| 02:15.39 | marlowe | matiasg - dont ask to ask - just ask |
| 02:15.39 | matiasg | ok thanx |
| 02:15.46 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 02:15.51 | matiasg | I got some pcs with softphones |
| 02:15.58 | HoopyCat | yay! squirrel has been removed, electricity is back. |
| 02:16.15 | matiasg | I got some of them working fine and two have an excellente outgoing sound quality |
| 02:16.29 | matiasg | but I hear the other party with a terrible choppy sound |
| 02:16.42 | matiasg | (just like talking in front of a fan) |
| 02:17.10 | matiasg | they are using the same soft phones |
| 02:17.24 | matiasg | the use the same account (one at a time of course) |
| 02:17.29 | matiasg | in the same * server |
| 02:17.32 | matiasg | same codecs |
| 02:17.44 | matiasg | I have even tried different sound cards |
| 02:17.55 | matiasg | (on the pcs which don't work fine) |
| 02:18.12 | TomL | what protocol? |
| 02:18.17 | matiasg | but the weird thing is: if I call an exten=>111,Playback(foo) |
| 02:18.22 | matiasg | that works fine |
| 02:18.31 | matiasg | (SIP & IAX both do the same) |
| 02:18.43 | TomL | is this all contained on a LAN, or are there WAN links involved? |
| 02:18.49 | matiasg | the voicemail prompt of course works fine |
| 02:18.55 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 02:19.05 | matiasg | all in a LAN dedicated to VoIP |
| 02:19.26 | matiasg | (lets say only *, pcs running softphones and ATAs) |
| 02:20.05 | matiasg | all using linksys rackable switches |
| 02:20.32 | JerJer[interop] | i'm sorry |
| 02:20.37 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 02:20.49 | matiasg | and one of the courious things is that the pcs not working fine are HP new pcs |
| 02:20.55 | *** join/#asterisk nDuff (~ccd@net-6621942-66.customer.corenap.com) |
| 02:21.00 | hermie | just because it has a little Cisco logo doesn't make it good |
| 02:21.13 | Qwell | matiasg: Is it that the soundcards just suck? |
| 02:21.13 | justnulling2 | i get disconnected after excactly 21 seconds every time i use asterisk, any ideas? |
| 02:21.16 | matiasg | not very good ones but at least they are new... |
| 02:21.25 | WilliamK | hermie, using the cheapest thing you can doesn't make it work well either =) |
| 02:21.42 | hermie | WilliamK: :) |
| 02:21.48 | matiasg | I know hermie but I'm trying to give as much info as I can... |
| 02:22.25 | matiasg | has anyone gone through something like this? |
| 02:22.27 | hermie | don't worry about us, we're just amused in our own little world |
| 02:23.07 | WilliamK | matiasg, try another switch stacked with that one and see if the problem goes away |
| 02:23.13 | ManxPower | MrBelvedr: PASTE the Dial that isn't working to the channel |
| 02:23.25 | matiasg | hermie: ok sorry thought you were talking 'bout the linksys stuff I wrote |
| 02:23.44 | Dr-Linux | matiasg: Pcs have static ips ? |
| 02:23.56 | MrBelvedr | thanx Manx |
| 02:23.57 | WilliamK | matiasg, cutting corners with voip isn't the best thing to do |
| 02:23.58 | MrBelvedr | exten => _1XXXXXXXXXX,1,DIAL(IAX2/thomasamiller@teliax/${EXTEN},20000,L(1000)) |
| 02:23.58 | matiasg | williamK: what do you mean? conect the pc and * to another switch? |
| 02:23.59 | WilliamK | =) |
| 02:24.15 | WilliamK | matiasg, hardware chipset problem on the nic probably |
| 02:24.20 | matiasg | yes they have static IPs |
| 02:24.38 | TomL | WilliamK: you beat me to it :P |
| 02:24.40 | MrBelvedr | the "L" option is being ignored. btw, "S" options also seem to be ignored |
| 02:24.44 | hermie | actually, looking at your problem, i'd say your networking gear is the problem somewhere matiasg |
| 02:24.46 | WilliamK | connect your workstations to another switch and plug the switch into the other |
| 02:24.51 | matiasg | williamK: what do you mean with cutting corners? (sorry for my english...) |
| 02:25.04 | ManxPower | MrBelvedr: *shrug* Maybe your timeout is too long. Try using 60 instead of 2000. |
| 02:25.13 | TomL | matiasg: do the HP PC's have built-in NICs? |
| 02:25.16 | WilliamK | matiasg, don't use the cheapest thing in the book, use managed products on switching gear, etc.. |
| 02:25.29 | MrBelvedr | exten => _1XXXXXXXXXX,1,DIAL(IAX2/thomasamiller@teliax/${EXTEN},1,L(1)) also fails though! |
| 02:25.31 | *** join/#asterisk asteriskDOTbz (~logger@telux.net) |
| 02:25.33 | asteriskDOTbz | <PROTECTED> |
| 02:25.35 | ManxPower | MrBelvedr: Also the docs COULD be wrong and the units might be seconds and not ms for the L switch. |
| 02:25.46 | ManxPower | MrBelvedr: What is the error message? |
| 02:25.53 | WilliamK | Linksys is also a consumer product, not a product most mid-sized companies use |
| 02:25.56 | ManxPower | nobody in their right mind would use 1. |
| 02:26.00 | ManxPower | Use 60 as I said. |
| 02:26.11 | MrBelvedr | there is not error message. the "L" is supposed to limit and it doesn't |
| 02:26.14 | MrBelvedr | i will try 60 |
| 02:26.15 | matiasg | TomL: yes they do... |
| 02:26.31 | nDuff | I'm trying to use speex for storing voicemail. I edited the formats line in voicemail.conf to read "format=speex|wav49|gsm|wav" -- but Asterisk complains "no such format 'speex'". I recompiled after installing libspeex, and /usr/lib/asterisk/modules/codec_speex.so exists -- is there anything else I need to do to get speex support working? |
| 02:26.50 | matiasg | they're managed switches (or at least they are in that category on their web site...) |
| 02:27.02 | ManxPower | nDuff: does "show translations" show speex as having numbers? |
| 02:27.19 | WilliamK | can you get into the switch with a web browser or telnet to manage each individual port? |
| 02:27.38 | ManxPower | MrBelvedr: If you can't get it working, post a message to the mailing list. |
| 02:27.40 | ManxPower | ~mailinglist |
| 02:27.41 | jbot | i guess 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/ |
| 02:27.42 | matiasg | yes it has a console administrative port |
| 02:27.55 | nDuff | ManxPower, yup. |
| 02:27.56 | matiasg | and a web management |
| 02:27.57 | WilliamK | matiasg, lock each port to 100/full duplex then |
| 02:28.05 | hermie | 3com or Allied Telesyn managed gear is pretty darn cheap but doesn't suck |
| 02:28.09 | WilliamK | that'll probably cure the problem |
| 02:28.32 | WilliamK | hermie, even netgear on switches works pretty well |
| 02:28.35 | WilliamK | just not on their routers |
| 02:28.44 | WilliamK | netgear routers suck more than linksys |
| 02:29.00 | Sedorox | ahaha |
| 02:29.07 | Sedorox | and I have a netgear right in front of me |
| 02:29.09 | Sedorox | I love them |
| 02:29.10 | *** join/#asterisk NewSole (david@69.156.51.222) |
| 02:29.17 | hermie | WilliamK: ask me about netgear network cards sometime |
| 02:29.22 | matiasg | williamK: thanks I'll try but what I don't really understand is why this happens only in the incoming audio, and why it doesn't happen with a Playback |
| 02:29.29 | WilliamK | hermie, love frying them eh? |
| 02:29.30 | WilliamK | =) |
| 02:30.07 | NewSole | ok I got a good Question for you.... I have a phone connected to FXS on TDM400P Card and I was wondering the best way to do a transfer |
| 02:30.34 | WilliamK | matiasg, all kinds of things happen when the 2 sides of a port won't synch up properly (full duplex / half duplex) mismatching |
| 02:30.50 | *** join/#asterisk macr ([U2FsdGVkX@83.32.94.0) |
| 02:30.53 | WilliamK | take your 100Mbps lan and turn it into a 56k modem |
| 02:31.00 | Sedorox | bbl |
| 02:31.14 | hermie | WilliamK: at my last job, we orded a few hundred boxes with FA312s... they randomly stopped working |
| 02:31.28 | ManxPower | NewSole: FLASH, just like for Centrex |
| 02:31.31 | hermie | WilliamK: so we had to RMA them _all_ |
| 02:31.35 | WilliamK | hermie, doesn't surprise me =) |
| 02:31.43 | WilliamK | I use Intel NICs on everything |
| 02:31.45 | nDuff | ManxPower, the only one that doesn't have numbers is g729 (which it looks like it can't convert to/from *anything*). |
| 02:31.48 | WilliamK | disable all the onboard's |
| 02:32.00 | hermie | WilliamK: I'm a big fan of the 3C905TX |
| 02:32.13 | WilliamK | I was a 3com fan till I had all kinds of problems |
| 02:32.16 | matiasg | williamK:ok i'll do it |
| 02:32.25 | WilliamK | dang nics randomly started dropping off the network |
| 02:32.48 | WilliamK | plus I get better performance with the intel's |
| 02:32.53 | hermie | WilliamK: i've been good with the classic 905s... we build firewall boxes for customer with em |
| 02:33.24 | WilliamK | hermie, we did that till I proved I could drop firewalls with 2.5Mbps of DNS traffic |
| 02:33.53 | hermie | WilliamK: oh, and Netgear sent us replacement cards from an unreleased model... a good 10% of which were bad. Have you ever had to RMA an RMA? |
| 02:34.15 | WilliamK | hermie, actually yes... Seagate harddrives |
| 02:34.20 | WilliamK | Barracuda series |
| 02:34.28 | hermie | really? newer ones? |
| 02:34.43 | WilliamK | 1997/98 |
| 02:34.49 | hermie | these days they're about the best on the market... 4 year warranty, 1+ million hr MTBF |
| 02:34.53 | WilliamK | shattered them within 24hrs of getting them |
| 02:34.55 | WilliamK | everytime |
| 02:35.08 | hermie | they've really improved |
| 02:35.26 | hermie | HW companies kinda go in cycles... USR used to have great wireless stuff |
| 02:35.46 | hermie | i've had to RMA their RMAs before too, but luckily is wasn't 350 NICs |
| 02:35.56 | WilliamK | I still use USR/3com Total Control chassis |
| 02:36.27 | hermie | their HW modems are good... but the wireless gear has gone south |
| 02:37.00 | MrBelvedr | damn |
| 02:37.13 | hermie | heh... http://i8.ebayimg.com/02/i/02/bf/cf/4d_1_b.JPG |
| 02:37.26 | *** join/#asterisk DrukenHME (~druken@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 02:37.54 | MrBelvedr | i can't imaging having to rma 350 boards |
| 02:38.11 | MrBelvedr | i would have to write a program just to do the paperwork |
| 02:38.18 | MrBelvedr | screw that |
| 02:38.20 | WilliamK | oh and WD drives |
| 02:38.32 | WilliamK | about 2 years ago we RMA'd several cases of them |
| 02:38.35 | hermie | WilliamK: well, WD went without saying :) |
| 02:38.46 | DrukenHME | if i had to rma 350 parts, i'd never buy that product again |
| 02:38.56 | WilliamK | we were replacing them by the cases |
| 02:38.57 | hermie | yeah... tell me about it |
| 02:39.12 | hermie | MrBelvedr: they just send us replacements by the case |
| 02:39.35 | hermie | MrBelvedr: and a FedEx airbill to return everything in the same box |
| 02:39.58 | hermie | the only WD stuff I'd buy is their "Special Editions" with the 8MB of cache |
| 02:41.20 | hermie | funny part of the whole Netgear story is that I now work for the company that sold us the machines with the bad NICs |
| 02:41.31 | MrBelvedr | personally i don't judge any company because of one bad product line |
| 02:41.42 | MrBelvedr | if they have a troublesome product line then avoid it |
| 02:42.10 | MrBelvedr | don't let one bad line spoil the whold bunch |
| 02:42.22 | DrukenHME | depends... |
| 02:42.40 | WilliamK | MrBelvedr, cheapest products on the market that are rated consumer, and companies with BAD customer service are the ones I avoid |
| 02:42.46 | WilliamK | avoid them with a passion |
| 02:43.32 | hermie | "we call it Customer Care (because service is what a bull does to a cow)" |
| 02:45.15 | tzanger | hahahahaha |
| 02:45.18 | matiasg | hermie: funny (and in account of being 23:15 in my country I need to read funny things...thanx) |
| 02:45.44 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 02:48.17 | *** join/#asterisk syslod (~yurplsl@65.114.0.198) |
| 02:49.17 | syslod | Hello |
| 02:50.11 | slePP | but too lazy to fix it :> |
| 02:50.13 | mishehu | syslod: seshu sent me the correct firmware, and got me the password to the phone. it's fairly decent for hte price. |
| 02:50.19 | file[laptop] | silly slePP |
| 02:50.27 | syslod | mishehu: As good as a poly? |
| 02:50.28 | slePP | actually, not so much SRV support |
| 02:50.33 | slePP | but rather the DNS caching that never expires |
| 02:50.51 | patdk | that is called bad dns config |
| 02:50.53 | mishehu | syslod: it's more like the ip300, but I've never actually used an ip300 before. also, it doesn't do poe |
| 02:50.55 | patdk | or stupid dns clients |
| 02:51.50 | syslod | mishehu: We've about got everything working on the poly and grandstream (Directories, paging, intercom). I'd like to pick one more phone to support. Does it have a backlit display. |
| 02:52.12 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:52.25 | syslod | mishehu: I am thinking about taking a proprietry mgcp phone and integrating cause nobody seems to have backlit displays. |
| 02:53.34 | DrukenHME | slePP: SRV ? |
| 02:53.54 | slePP | records that give you multiple options for one host, basically |
| 02:54.24 | DrukenHME | mmm, k |
| 02:55.24 | slePP | my NAT is so broken... |
| 02:55.36 | syslod | Anyone noticed that app_voicemail seems to be tring to send message.WAV now rather than message.wav? It is sending a blank file all the sudden. |
| 02:56.06 | MrBelvedr | show hints |
| 02:56.14 | MrBelvedr | oops |
| 02:56.35 | mishehu | syslod: this is better than the barbietones |
| 02:56.46 | mishehu | the plastic doesn't feel so cheap |
| 02:56.47 | mishehu | heh |
| 02:57.31 | mishehu | syslod: mine's backlit when ringing of off-hook |
| 02:57.35 | syslod | mishehu: I'm happy with the polys but the no backlit thing seems to be an issue. |
| 02:57.48 | syslod | Does it lite up when in use? |
| 02:58.02 | matiasg | williamK: I have fixed the card speed, disabled the on board card, used a 3com card instead... now I'm going to try to use another switch |
| 02:58.09 | matiasg | stacked with the linksys |
| 02:58.18 | syslod | Take a look at the pana 7600 series. |
| 02:58.23 | matiasg | williamK: of course no luck yet.... |
| 02:58.24 | mishehu | syslod: off-hook would be in use ;-) also, i only seem to get numeric cid. |
| 02:58.57 | syslod | Thats sounds promising beside the numeric cid. Does it have NOTIFY events for autoanswer etc? |
| 02:59.05 | Jim^^ | hmm I'm still on v1-0_stable, much point in updating? |
| 02:59.32 | *** join/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
| 03:02.47 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 03:02.51 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 03:04.53 | blitzrage | Damin: hey, I don't have your email, but I had to switch my public key today |
| 03:05.02 | blitzrage | Damin: in regards to dundi |
| 03:05.11 | blitzrage | Damin: I also need a link to your public key again... |
| 03:05.24 | *** join/#asterisk stillbourne (~stillbour@c-67-161-137-18.hsd1.co.comcast.net) |
| 03:06.16 | blitzrage | *gasp* :) |
| 03:06.16 | blitzrage | I have to go and beat my roommate at soccer now |
| 03:06.16 | blitzrage | back in a bit |
| 03:06.16 | file[laptop] | have fun |
| 03:06.16 | blitzrage | will do :) |
| 03:06.56 | matiasg | williamK: final test, connected the pc to * box with a inverted patch, keeps doing the same.... |
| 03:17.58 | *** join/#asterisk bjohnson (~bjohnson@ip209-179.tor.istop.com) |
| 03:19.08 | *** join/#asterisk moy (~moy@201.138.195.37) |
| 03:19.47 | matiasg | ok I'm going home bye everyone |
| 03:20.40 | newl | $HOME is where the $PROMPT is. |
| 03:23.37 | *** part/#asterisk moy (~moy@201.138.195.37) |
| 03:24.28 | *** join/#asterisk Rick_Hunter (~rhunter@03-106.008.popsite.net) |
| 03:27.57 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 03:28.05 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 03:32.36 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 03:34.47 | *** join/#asterisk Inv_arp (junya@adsl-3-255-42.mia.bellsouth.net) |
| 03:35.48 | PBXtech | anyone know what bkw is working on with the sms stuff? |
| 03:37.02 | PBXtech | Laughing Out Loud |
| 03:37.16 | PBXtech | oop |
| 03:40.46 | Dr-Linux | how i'll define, if someone dial "111" from outside, he direct listen IVR ? |
| 03:41.29 | *** join/#asterisk Rick_Hunter (~rhunter@03-106.008.popsite.net) |
| 03:44.35 | *** join/#asterisk asteriskDOTbz (~logger@telux.net) |
| 03:44.37 | asteriskDOTbz | <PROTECTED> |
| 03:46.19 | MikeJ[Laptop] | PBXtech, I know what he is working on |
| 03:46.39 | PBXtech | whats that |
| 03:46.46 | MikeJ[Laptop] | with sms stuff |
| 03:47.00 | PBXtech | yea and.. :) |
| 03:47.22 | MikeJ[Laptop] | not ready yet... he was saying he was going to submit it |
| 03:47.36 | MikeJ[Laptop] | wait and see or ask him, not really my place to say |
| 03:47.36 | PBXtech | k |
| 03:52.53 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 03:55.38 | bjohnson | Dr-Linux: make an exten for 111 that calls voicemil |
| 03:56.11 | bjohnson | MikeJ[Jayden]: thanks for providing all that info |
| 04:13.35 | PBXtech | do many people use dundi ? |
| 04:14.17 | *** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 04:14.45 | bkw_ | file[laptop], |
| 04:14.46 | bkw_ | wakeup |
| 04:14.51 | file[laptop] | yessssss? |
| 04:15.01 | bkw_ | msg me your cell again |
| 04:15.02 | bkw_ | please |
| 04:15.30 | file[laptop] | yay 1X |
| 04:16.14 | file[laptop] | website test received |
| 04:16.56 | DEEZED | i used to have cricket |
| 04:19.36 | MrBelvedr | for some reason all calls are ending after 24 seconds. I am using teliax and voipjet. With both providers it ends around 20 seconds |
| 04:19.44 | MrBelvedr | what would likely be causing that |
| 04:19.52 | *** join/#asterisk shodan (~shodan@216.113.99.249) |
| 04:20.07 | ManxPower | MrBelvedr: AbsoluteTimeout set? |
| 04:20.19 | ManxPower | MrBelvedr: Did you get your L() option working? |
| 04:20.20 | MrBelvedr | what file sets that? |
| 04:20.28 | MrBelvedr | no i never got L working |
| 04:20.34 | ManxPower | MrBelvedr: You would have to call it in your dialplan if you wanted it set. |
| 04:20.35 | MrBelvedr | did you try it on your box? |
| 04:20.40 | ManxPower | MrBelvedr: No. |
| 04:20.50 | ManxPower | MrBelvedr: What does the console show? That does DIALSTATUS show? |
| 04:20.50 | MrBelvedr | i don't ahve AbsoluteTimeout in my dialplan anywhere |
| 04:21.12 | ManxPower | That == What |
| 04:21.17 | *** join/#asterisk linsys (~non@70-57-11-107.dnvr.qwest.net) |
| 04:21.37 | MrBelvedr | should I put DIALSTATUS in my dial plan? or just show you what the cli says? |
| 04:21.57 | linsys | Can someone here suggest a good free softphone? Or someone familiar with X-Lite? |
| 04:22.06 | ManxPower | You would need something like Noop(DIALSTATUS=${DIALSTATUS} as the priority after the Dial line |
| 04:22.19 | ManxPower | MrBelvedr: If it's more than 2 lines, use pastebin.ca to paste stuff. |
| 04:22.24 | MrBelvedr | k |
| 04:22.32 | tzanger | two lines? |
| 04:22.37 | tzanger | that's a little tight don't you think? |
| 04:22.50 | tzanger | I thought the standard was 4-5 lines |
| 04:22.50 | PBXtech | where is asterlinks NOC located? |
| 04:23.15 | ManxPower | tzanger: *shrug* |
| 04:23.21 | newl | linsys: kphone, sjphone, linphone, x-lite, there's probably others. |
| 04:23.42 | MrBelvedr | Manx, here is the CLI output http://pastebin.ca/8709 |
| 04:24.16 | MrBelvedr | for some reason it thinks that nobody picked up |
| 04:24.19 | ManxPower | MrBelvedr: Looks like you have a firewall provlem. |
| 04:24.30 | MrBelvedr | the call is placed fine thouhg |
| 04:24.34 | ManxPower | I use teliax and it works fine. |
| 04:24.50 | MrBelvedr | my cell phone rings and i am able to talk and hear |
| 04:25.01 | ManxPower | MrBelvedr: "max retries exceeded" means "the far side stopped responding" |
| 04:25.08 | MrBelvedr | only problem is that the call ends after around 24 seconds |
| 04:25.11 | linsys | Well I have asterisk installed and when I run it with /usr/sbin/asterisk -vvvgc I get a command line so it seems to be working I don't see any errors |
| 04:25.11 | ManxPower | MrBelvedr: try notransfer=yes in iax.conf |
| 04:25.29 | MrBelvedr | ok will do now |
| 04:25.30 | MrBelvedr | thx |
| 04:25.34 | *** join/#asterisk bjohnson (~bjohnson@ip141-172.dsl.istop.com) |
| 04:25.52 | ManxPower | the not ransfer option can be tricky. |
| 04:25.58 | MrBelvedr | btw, that CLI is from my voipjet account. I will paste my teliax stuff in a bit |
| 04:26.23 | ManxPower | MrBelvedr: I won't use VoipJet so I can't really help you with them. |
| 04:26.24 | MrBelvedr | what does notransfer = yes do? |
| 04:26.34 | linsys | but when I try and get x-lite to connect, it even shows connected in the display, I don't see ANYTHING in on the CLI and when I try and dial an extension I have setup or even the voice mail box rerevial system I setup, I get a fast busy after like 20-30 seconds.. I even try tcpdump -n ip and port 5060 and I don't see anything... |
| 04:26.42 | linsys | so I'm not beliveing that it's connected.. |
| 04:26.44 | ManxPower | MrBelvedr: It prevents IAX2 transfers from happening |
| 04:26.45 | linsys | even though it says so.. |
| 04:27.18 | ManxPower | NAT can cause problems with IAX2 transfers. |
| 04:27.22 | ManxPower | Not common, but it can happen. |
| 04:27.59 | newl | linsys: If the config requester doesn't keep popping up, it in theory should be connected, though asterisk will show it register if in fact it's happening. |
| 04:28.17 | linsys | I should see a register message in the CLI? |
| 04:28.24 | newl | you should |
| 04:28.44 | MrBelvedr | I do have NAT |
| 04:28.52 | newl | linsys: try -cvvvd instead for more output |
| 04:28.54 | MrBelvedr | i will test it all out brb |
| 04:29.08 | linsys | and the config requester doesn't keep poping up.. I see "Logged In - Enter Phone Number" "Your Number is 2000" |
| 04:29.12 | linsys | k |
| 04:29.13 | Jim^^ | anyone noticed that voicemail emailing of wav49 seems to be broken? |
| 04:29.15 | linsys | I will try that.. |
| 04:29.48 | linsys | I do see this |
| 04:29.49 | linsys | Apr 2 01:24:00 WARNING[15408]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled |
| 04:29.52 | linsys | is that an issue? |
| 04:29.54 | newl | nope |
| 04:29.55 | file[laptop] | no. |
| 04:30.30 | ManxPower | linsys: Not normally, but chan_sip DOES have some issues with ip resolution. Make sure your machine/ip is in /etc/hosts |
| 04:31.05 | shodan | can I use a wildcard x100p as a fax receiver ? (and sender ?) (and as a 56k modem ?) |
| 04:31.31 | ManxPower | shodan: A1: Yes. A2: It's a LOT of work to set it all set up and it still may not work. |
| 04:31.38 | Qwell | using an x100p as a modem...thats a novel idea. heh |
| 04:31.58 | PBXtech | i still have some probs with spand pre11 |
| 04:32.07 | PBXtech | not as bad |
| 04:33.03 | linsys | Hmmm fixed /etc/hosts (this box was taken off another network...) closed my soft phone and the phone says it loggs back in.. but I don't see anything inside of the CLI |
| 04:33.24 | Qwell | Dr-Linux: Don't message me |
| 04:33.39 | PBXtech | or me |
| 04:33.43 | file[laptop] | or me |
| 04:33.45 | PBXtech | :/ |
| 04:33.46 | Qwell | ... |
| 04:33.51 | newl | (recap question/statement from half a moon ago) Well, this isn't good. B number has active diversion to C number. A party calls B number which gets diverted to C number. A party gets CDR recorded for the diversion, which is totally incorrect handling. The B number should get the diversion recorded against it. If anyone else can duplicate this behavior, I'll file a bug. Alternatively, if someone has another method of telling asterisk t |
| 04:33.52 | newl | hat the B number diverting should take the hit for the CDR, that'd be better yet. |
| 04:33.53 | linsys | I just have basic settings in my X-lite like "Enabled Yes" Display Name " |
| 04:33.59 | Qwell | assbag |
| 04:34.00 | ManxPower | linsys: Maybe the phone is connecting to some other service |
| 04:34.13 | Dr-Linux | file[laptop]: ? |
| 04:34.17 | linsys | you know I did think that was a possibiluty |
| 04:34.20 | newl | like a stun server perhaps. |
| 04:34.21 | file[laptop] | pre-warning |
| 04:34.25 | Qwell | heh |
| 04:34.30 | ManxPower | newl: Using IAX? |
| 04:34.43 | linsys | it seems to say in the begining discovering firewall |
| 04:34.44 | file[laptop] | Dr-Linux: Don't message me *G* |
| 04:34.44 | PBXtech | got a Q ask in channel |
| 04:34.54 | file[laptop] | see? there's the actual warning |
| 04:34.59 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
| 04:35.04 | newl | ManxPower: nope, sip. One asterisk, one GS 101, one x-ten all running on the same subnet, no firewall. |
| 04:35.05 | shodan | ManxPower, that's ok , as long as there's a remote possibility I don't mind working for it |
| 04:35.10 | linsys | which is odd because the asterisk box is on the same network as the X-Lite box.. |
| 04:35.22 | ManxPower | newl: weird |
| 04:35.37 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 04:35.48 | Dr-Linux | file[laptop]: what msg i sent you ? :P |
| 04:35.51 | linsys | All I should have to configure is the info in "Proxy" in the config |
| 04:35.53 | file[laptop] | haha |
| 04:35.56 | linsys | right? |
| 04:36.02 | newl | ManxPower: I thought so too. It's almost akin to the CLIP issue but instead it's CDR based, not presentation based. |
| 04:36.48 | linsys | Like OutBound Proxy I see that under the Proxy config and network config... but I left it blank... the only proxy I filled in was under SIP Proxy for the Option SIP Proxy |
| 04:36.54 | linsys | which is the IP address of the asterisk box.. |
| 04:37.00 | *** join/#asterisk bparker (bparker@cable-71-8-65-183.mtv.al.charter.com) |
| 04:38.17 | Dr-Linux | i'm using softphones, now i gonna add X100P , what things need to be change ? |
| 04:39.17 | linsys | I also see this message in my asterisk message log Unable to load config iax.conf but I'm using SIP not iax.. should I just remove that config? |
| 04:39.25 | linsys | does one take presidence over the others |
| 04:39.31 | linsys | or can they both exist? |
| 04:39.37 | nDuff | Any word on whenbouts there'll be a stable release incorporating RealTime support? |
| 04:40.48 | file[laptop] | who needs asterisk help and will pay? eh? EH? |
| 04:41.03 | newl | linsys: I've got the following set in sip proxy for my x-lite client, username, auth user, password, domain/realm sip proxy, outbound proxy. I've also got Transmit Silence set to Yes (advanced->audio->silence) |
| 04:41.15 | Dr-Linux | file[laptop]: lolzz |
| 04:41.26 | file[laptop] | I've gotta pay Fedex somehow :p |
| 04:41.27 | linsys | I don't have domain/realm set at all.. |
| 04:41.31 | newl | file thinks he's a canook |
| 04:41.37 | ManxPower | file[laptop]: FedEx them a check? |
| 04:41.37 | linsys | and no outbound proxy |
| 04:41.47 | newl | linsys: I've got them set to the ip of the asterisk server here. |
| 04:41.50 | file[laptop] | ManxPower: doesn't work like 'dat |
| 04:41.57 | newl | file[laptop]: what? no .ca? :) |
| 04:42.00 | bkw_ | ok anyone have cricket or some sane provider that don't dink with the crap |
| 04:42.09 | linsys | which the domain/realm or the outbound proxy? |
| 04:42.12 | file[laptop] | LOL |
| 04:42.17 | file[laptop] | bkw_: given up on Telus? :( |
| 04:43.50 | linsys | newl: Do you see any log in info in the CLI when your softphone logs in? |
| 04:46.13 | bkw_ | file[laptop], I have no clue how to make this work with telus |
| 04:46.17 | ManxPower | newl: If the phone does not show up (with the correct IP address) in "sip show peers" then it's NOT registering |
| 04:46.24 | file[laptop] | bkw_: lol they stumped you? that's depressing |
| 04:46.38 | bkw_ | I just don't feel like dickin with it |
| 04:46.55 | file[laptop] | aww no dickin' around? |
| 04:47.49 | linsys | Also when I run sip show peers I see both of my extensions, but both hosts say "Unspecified" on the asterisk box.. but like I said my Softphone says it's connected |
| 04:48.00 | linsys | that measn it didn't connect, right? |
| 04:48.06 | ManxPower | linsys: The Softphone LIES |
| 04:48.17 | mithro | anyone know why FXS adapters are so expensive? |
| 04:48.30 | ManxPower | ~google site:lists.digium.com x-lite register |
| 04:48.33 | Qwell | mithro: They aren't |
| 04:48.52 | Qwell | mithro: How much would each avaya extension cost? |
| 04:48.54 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 04:49.13 | newl | linsys: yep, -- Registered SIP '1314' at 10.0.0.100 port 5061 expires 1800 |
| 04:49.17 | shodan | mithro, because people will buy them at that price |
| 04:49.34 | mithro | avaya? |
| 04:49.46 | ManxPower | or Cisco or Nortel or AT&T or or or |
| 04:50.17 | mithro | why are FXS adapters more expensive then FXO? i can get FXO for like $8 US, arn't they technically very similar? |
| 04:50.34 | mithro | Qwell: links to avaya? |
| 04:50.38 | ManxPower | mithro: FXS devices are somewhat more complicated. |
| 04:50.50 | ManxPower | mithro: And you won't be able to get FXO for $8 soon either. |
| 04:51.01 | linsys | I see 2000/2000 (Unspecified) D 255.255.255.255 0 Unmonitored |
| 04:51.14 | linsys | for sip show peers |
| 04:51.16 | ManxPower | linsys: Unspecified means THE PHONE DIDN'T REGISTER. |
| 04:51.19 | mithro | ManxPower: why? |
| 04:51.22 | Qwell | ManxPower: oh? |
| 04:51.25 | linsys | right, I got that.. |
| 04:51.32 | ManxPower | mithro: Intel stopped making the chip that those cards use. |
| 04:51.38 | shodan | they are expensive considering the parts list , considering that are AF DAC+ADC with a DC to DC converter |
| 04:52.08 | ManxPower | linsys: see the google links jbot pasted. |
| 04:52.10 | linsys | I also see this in my X-Lite log |
| 04:52.10 | linsys | SIP: 192.168.0.100:5060 |
| 04:52.10 | linsys | RTP: 192.168.0.100:8000 |
| 04:52.11 | linsys | NAT: 70.57.11.107 |
| 04:52.18 | shodan | ManxPower, they're made by ambient , is it intel in fact who's making them ? |
| 04:52.23 | linsys | l |
| 04:52.25 | linsys | k |
| 04:52.29 | shodan | (ambient md3200) |
| 04:52.34 | ManxPower | shodan: I had thought Intel bought Amdient |
| 04:52.53 | shodan | I'll look it up |
| 04:53.03 | mithro | ManxPower: dang |
| 04:53.38 | ManxPower | mithro: FXO is so cheap because modems are basically an FXO device, so the parts and chips are cheap. |
| 04:53.57 | ManxPower | At least in the PCI world. |
| 04:54.02 | mithro | isn't it just a Soundcard with an isolation device? |
| 04:54.03 | shodan | oh ok , it did feb2000 |
| 04:54.21 | shodan | ManxPower, but a fxo is a simple device anyway , no ? |
| 04:54.34 | ManxPower | mithro: There is DAC, DSP, filters, etc. |
| 04:54.42 | ManxPower | shodan: As I understand it, yes. |
| 04:54.44 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.res.rr.com) |
| 04:54.54 | nine76 | hello all |
| 04:55.01 | linsys | I don't have a register line in my sip.conf |
| 04:55.34 | ManxPower | linsys: register => in sip.conf is for registering to remote servers. |
| 04:55.45 | linsys | ok... that's what I thought |
| 04:55.49 | shodan | and POTS can't go over 4khz right ? so 8khz DAC/ADC are all you need |
| 04:55.51 | linsys | that's what those articles are about.. |
| 04:56.13 | ManxPower | shodan: that gets into the electronics geek realm and I'm not one of those. |
| 04:56.22 | nine76 | If anyone has played with areskicc I'd appreciate them telling me how to get past the last steps from wiki... i.e. http://pastebin.ca/8710 please,and thx:-/ |
| 04:57.08 | shodan | k , I'm playing with pic MCUs right now , I'll try making a 5$ fxs when I master them |
| 04:57.21 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 04:57.24 | ManxPower | shodan: best of luck with it. |
| 04:58.24 | florz | shodan: If you wanna make it really cheap, would something != PIC be better? =:-) |
| 04:58.34 | shmaltz | anybodoy here that knows or has a good gui to configure asterisk, that is based on the text files? |
| 04:58.55 | mithro | so if i wanted cheap FXS ports what would you recommend? |
| 04:59.47 | Dr-Linux | ManxPower: where i can find the file, the welcome greeting IVR ? |
| 05:00.09 | Dr-Linux | ManxPower: default file path ? |
| 05:00.57 | florz | shodan: erm, +n't |
| 05:03.03 | shmaltz | anybodoy here that knows or has a good gui to configure asterisk, that is based on the text files? |
| 05:03.23 | bjohnson | mithro: Sipura SPA 2000 |
| 05:03.29 | bjohnson | shmaltz: doesn't exist IMO |
| 05:03.45 | bkw_ | what what what? |
| 05:03.51 | bkw_ | someone asking questions? |
| 05:03.51 | bjohnson | Dr-Linux: look in the sample conf files |
| 05:04.02 | MikeJ[Laptop] | no questions here.... |
| 05:04.05 | MikeJ[Laptop] | just answers |
| 05:04.06 | shmaltz | bjohnson, of course there is at least one that exists, its made by thirdlane.com |
| 05:04.22 | shmaltz | bkw_, nybodoy here that knows or has a good gui to configure asterisk, that is based on the text files? |
| 05:04.31 | bkw_ | zero |
| 05:04.33 | MikeJ[Laptop] | my answers, in no particular order are: rtfw, 2pi, and 42 |
| 05:04.35 | bjohnson | shmaltz: go for it .. ps .. you said good |
| 05:04.42 | mithro | bjohnson: dang, the Sipura SPA 2000 is still a bit pricey for me :/ |
| 05:04.52 | bjohnson | mithro: forget about voip |
| 05:05.12 | *** join/#asterisk uncrfe (~uncrfe@69.145.65.248) |
| 05:05.13 | shmaltz | bjohnso, it's actualy a good product, but it is not context aware, and it has to be context aware |
| 05:05.16 | MikeJ[Laptop] | shmaltz, phpconfig is a gui based upon text files. |
| 05:05.35 | MikeJ[Laptop] | all it really is is a web based text file editor tho |
| 05:05.47 | mithro | why are there no PCI FXS cards apart from the diguim one? |
| 05:05.59 | bjohnson | hehe .. kate is a good gui text file editor |
| 05:06.21 | shmaltz | MkikJ, where can I get phpconifg? |
| 05:06.47 | shmaltz | MikeJ, Well, then webmin is much better |
| 05:07.01 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 05:07.03 | bjohnson | mithro: because most if the big boys play with channel banks with T1 interfaces |
| 05:07.12 | MikeJ[Laptop] | shmaltz, see the first answer above ;) |
| 05:07.14 | shmaltz | ok i'll rephrae my question |
| 05:07.30 | shmaltz | anybody here want to make some money? |
| 05:07.46 | MikeJ[Laptop] | yes |
| 05:07.47 | bjohnson | shmaltz: consider that most of the people here find the guis too restirctive and edit the conf files by hand |
| 05:08.03 | shmaltz | bjohnson, thats what I do |
| 05:08.13 | shmaltz | how ever I must get a gui for a client |
| 05:08.28 | shmaltz | MikeJ, can you explain why you qualify? |
| 05:08.34 | bjohnson | to do what? |
| 05:08.44 | MikeJ[Laptop] | I never said I qualified |
| 05:08.49 | shmaltz | bj,manage asterisk as a PBX |
| 05:08.57 | MikeJ[Laptop] | you just asked if anyone wanted some money |
| 05:09.00 | shmaltz | so why did you say yes |
| 05:09.07 | MikeJ[Laptop] | see ^^ |
| 05:09.09 | bjohnson | wouldn't you? |
| 05:09.14 | shmaltz | ok , you got me |
| 05:09.18 | shmaltz | ;p |
| 05:09.19 | MikeJ[Laptop] | :) |
| 05:09.23 | linsys | It seems if my X-Lite keeps wanting to talk with my firewall.. |
| 05:09.28 | MikeJ[Laptop] | it depends, what do you want |
| 05:09.35 | linsys | when I sniff the network all I see is traffic 02:03:06.597970 IP 192.168.0.2.1900 > 239.255.255.250.1900: UDP, length 312 |
| 05:09.37 | linsys | like that.. |
| 05:09.45 | shmaltz | so let me rephrase again |
| 05:09.48 | linsys | I think those are the UDP keep alives from X-Lite.. |
| 05:10.18 | uncrfe | I have a question: I'm investigating setting up a PBX for a nonprofit. They've been quoted $12000 for a pbx system from the phone co. From what I've looked at, I should be able to get a t1 in, connect from the demarc to a te110p in a server, connect that server's netcard to a switch that has the ip phones connected to it, and have (with configuration) a running pbx w/voicemail and DID. Am I correct? |
| 05:10.19 | shmaltz | anybody here that qualifies to program such a project is intersted in creating a gui for * that is based on the text files? |
| 05:10.28 | linsys | RECEIVE << 64.69.76.23:3478 I also see that in the X-Lite log... not sure why.. I don't even know if what that IP is.. |
| 05:10.34 | shmaltz | for money of course |
| 05:10.57 | MikeJ[Laptop] | uncrfe, yes |
| 05:11.00 | iq | shmaltz, you couldn't ffffind any GUI? |
| 05:11.11 | shmaltz | iq, well could you? |
| 05:11.24 | iq | shmaltz, to create extensions, dial plans, etc. ? |
| 05:11.31 | shmaltz | yep, iq |
| 05:11.48 | MikeJ[Laptop] | shmaltz, what are you looking for? |
| 05:11.57 | iq | shmaltz, there are few out there (with source). I used one about 4 months back |
| 05:12.26 | shmaltz | gui to create configure: |
| 05:12.28 | shmaltz | zap, sip, extensions.conf, musiconhold, and should be context aware |
| 05:12.32 | shmaltz | iq, name please? |
| 05:12.51 | bjohnson | MikeJ[Jayden]: he'll know it when he sees it |
| 05:12.51 | iq | shmaltz, I dont remember. Let me do google for you |
| 05:12.56 | bjohnson | try AMP |
| 05:12.58 | shmaltz | as well as voicemail.conf |
| 05:13.07 | MikeJ[Laptop] | AMP does that, but is probably too restrictive |
| 05:13.07 | shmaltz | amp doesn't work for me, sorry |
| 05:13.19 | bjohnson | they are all too restrictive |
| 05:13.42 | bjohnson | first step in making a gui .. make it simpler by removing options |
| 05:13.52 | MikeJ[Laptop] | type up detailed specs of what you want it to look like, and post it to the wiki with a bounty |
| 05:14.03 | bjohnson | detailed |
| 05:14.10 | MikeJ[Laptop] | send an e-mail to the dev list anouncing the bounty |
| 05:14.12 | mithro | so overall atm Sipura SPA-2000 is the cheapest FXS solution? |
| 05:14.22 | bjohnson | mithro: no |
| 05:14.34 | uncrfe | MikeJ[Laptop], excellent. I've found the default docs, and read through them. They seem to be focussed on creating such a setup with a single inbound analog line. I haven't found any howto / guide on how to set up for an incoming t1 of lines |
| 05:14.34 | bjohnson | mithro: one of the cheapest per port .. it's a 2 port fxs |
| 05:14.40 | iq | http://www.tryvibe.com |
| 05:14.54 | mithro | bjohnson: yeah i meant cheapest per port |
| 05:14.56 | bjohnson | uncrfe: same idea |
| 05:15.03 | bjohnson | ~docs |
| 05:15.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 05:15.05 | uncrfe | cool..... |
| 05:15.06 | mithro | (well cheapest for a 2 or 4 port configuration) |
| 05:15.16 | bjohnson | uncrfe: many, many, many, installs of T1 systems |
| 05:15.36 | uncrfe | any available demo confs for such? |
| 05:15.38 | bjohnson | mithro: also cheapest per port for a 1 or 3 port systems |
| 05:15.52 | bjohnson | uncrfe: likely yes .. on the wiki |
| 05:16.08 | MikeJ[Laptop] | uncrfe, t1's send dnis digits, same kinda deal, just instead of s ext, you use the dnis digits.... it's really pretty easy once you get in and start playing with it |
| 05:16.16 | bjohnson | uncrfe: the only difference between a T1 and an analogue line in is the zapata.conf |
| 05:16.28 | MikeJ[Laptop] | I think there are some demo confs on the wiki |
| 05:16.28 | MikeJ[Laptop] | ~rtfw |
| 05:16.29 | jbot | [rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 05:16.32 | bjohnson | and the extensions.conf is likely a little bigger with more phones |
| 05:16.33 | MikeJ[Laptop] | :) |
| 05:16.36 | uncrfe | one last question: is it realistic (after reading through all the docs and the book) to expect myself to be able to get a setup running (at least to the point of DID / vm) in a week or so? |
| 05:16.36 | iq | shmaltz, : try this sourceforge.net///////projecttttttts/astguiclient |
| 05:16.47 | bjohnson | uncrfe: no |
| 05:16.47 | uncrfe | (nods at the wiki comment) |
| 05:16.54 | iq | shmaltz, this is not the one I used thoughhhhhhhhhhh |
| 05:17.12 | bjohnson | uncrfe: well .. if you work at it 100% |
| 05:17.27 | bjohnson | uncrfe: steep learning curve until you understand the basics |
| 05:17.33 | bjohnson | uncrfe: btw .. vm is easy |
| 05:17.40 | bjohnson | once you get the other stuff working |
| 05:17.43 | uncrfe | yeah....looks easy from the docs |
| 05:17.43 | shmaltz | iq, its no good, i tried it |
| 05:17.52 | uncrfe | I would be. It would be my #1 priority (after sleep) to have it running |
| 05:18.05 | bjohnson | you might have to move sleep to #2 |
| 05:18.22 | *** join/#asterisk vlan (~iq@207-224-100-44.omah.qwest.net) |
| 05:19.06 | vlan | shmaltz: oh okay. Let me do more google :) |
| 05:19.15 | uncrfe | lots of experience with linux in general, w/phone stuff, but not with pbx stuff. The basics (if by basics you mean the stuff in docs) seem pretty straightforward |
| 05:19.18 | shmaltz | vlan, thanks |
| 05:21.05 | uncrfe | ok, found the configs on the wiki |
| 05:22.10 | vlan | shmaltz: nothing u like here: http://www.voip-info.org/wiki-Asterisk+GUI |
| 05:22.37 | shmaltz | vlan, you think I haven't wasted 2 weeks on that page? |
| 05:23.00 | vlan | shmaltz: I believe you :) ...sorry I can't recall the one I used :( |
| 05:24.15 | shmaltz | it's OK, but I can't get it why I can't find anybody iterested in this for money |
| 05:25.45 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 05:26.21 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 05:26.23 | uncrfe | bjohnson, MikeJ[Laptop] -- Thank you, off to do more reading |
| 05:27.00 | *** join/#asterisk mkhan (~mkhan@dsl092-066-137.bos1.dsl.speakeasy.net) |
| 05:28.39 | mkhan | hello.. I just put my TDM400P on my computer.. |
| 05:29.13 | mkhan | it was automatically detected as hisax .. |
| 05:29.21 | mkhan | what should I do .. for Zaptel |
| 05:29.38 | ManxPower | install zaptel |
| 05:30.30 | uncrfe | actually, one last question for bjohnson: what if I had time to config everything -but- the incoming t1 ahead of that week? |
| 05:30.58 | uncrfe | (basically, they sign lease may 1, I need to have phones running in 1-1.5wk) |
| 05:31.07 | mkhan | why is Zaptel for.. isn't it the driver of the card? |
| 05:31.34 | uncrfe | but I could get the hardware / sw running ahead of time, I think |
| 05:33.33 | Dr-Linux | mkhan: singa chal day |
| 05:33.50 | Dr-Linux | mkhan: kya hall hai beta ? |
| 05:36.04 | shmaltz | anybody here that qualifies to program such a project is intersted in creating a gui for * that is based on the text files? if you do please reply, if you end up getting this job, you will be paid |
| 05:37.53 | uncrfe | bjohnson? |
| 05:38.07 | florz | shmaltz: some more spec would be helpful, I guess |
| 05:38.32 | *** join/#asterisk shodan (~shodan@216.113.99.157) |
| 05:38.39 | shmaltz | gui to create configure: |
| 05:38.41 | shmaltz | zap, sip, extensions.conf, musiconhold, voicemail.conf, and should be context aware |
| 05:38.57 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 05:39.28 | drooth | anyone here run * with pfsense or monowall? |
| 05:40.57 | linsys | Ok, I got my softphone to log in using SJPhone |
| 05:41.04 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 05:41.35 | *** join/#asterisk ard (~ard@2001:7b8:32d:0:20c:6eff:fe18:d11f) |
| 05:41.48 | linsys | however when I try and dial an extension I setup I immediatly get a message saying "Number Not Available" "Call Rejected: 503 Service Unavailable" |
| 05:41.53 | linsys | any ideas on that message? |
| 05:45.45 | MrBelvedr | linsys are you sure you have the right IP or dns in your iax.conf |
| 05:45.57 | MrBelvedr | it looks like you are trying to connect to the wrong server |
| 05:46.00 | MrBelvedr | just a guess |
| 05:46.15 | MrBelvedr | are you able to ping the voip provider from your * box |
| 05:46.53 | shmaltz | anybody here that qualifies to program such a project is intersted in creating a gui for * that is based on the text files? if you do please reply, if you end up getting this job, you will be paid |
| 05:46.55 | shmaltz | gui to create configure: |
| 05:46.57 | shmaltz | zap, sip, extensions.conf, musiconhold, voicemail.conf, and should be context aware |
| 05:47.22 | linsys | MrBelvedr: I'm just trying to call a local extension |
| 05:47.42 | linsys | I don't have this connected to my phone line yet.. I'm going to use an x100P to connect to my analogue line |
| 05:47.59 | linsys | but I have a 2 extensions setup in my sip.conf |
| 05:48.08 | linsys | 2000 (which I can log into finally) and 2001 |
| 05:48.16 | linsys | which I set to go to vmail if the line was busy |
| 05:48.27 | linsys | I also have set 2999 for checking vmail |
| 05:48.40 | linsys | my iax.conf is empty |
| 05:53.39 | *** join/#asterisk file[laptop] (~file@mctn1-3451.nb.aliant.net) |
| 05:54.58 | shmaltz | test |
| 05:55.04 | shmaltz | ~helo |
| 05:55.06 | jbot | it has been said that helo is the first command issued during smtp |
| 05:55.12 | Supaplex | as far as ATA's go, are Sipura's the most $$$ for the buck? I need 802.3 (no wifi) |
| 05:55.14 | linsys | see when I dial 2999 I get that message, here is my line in my extensions.conf exten => 2999,1,VoicemailMain(${CALLERIDNUM}) |
| 05:55.16 | linsys | exten => 2999,1,VoicemailMain(${CALLERIDNUM}) |
| 05:55.20 | *** join/#asterisk Derkommissar (~alberto@fl-southhub-u1-c6-0a-174.miamfl.adelphia.net) |
| 05:55.27 | linsys | I should be able to just dial 2999 right? |
| 05:55.39 | Derkommissar | does anyone here have expirience with chan_unicall |
| 05:55.47 | Derkommissar | or the R2 Libs |
| 05:55.52 | Derkommissar | :( |
| 05:56.02 | Derkommissar | I been going at it for hours. |
| 05:59.44 | mkhan | I would like to rpm from zaptel source |
| 05:59.58 | mkhan | is there any spec file available |
| 06:04.59 | Derkommissar | you can just compile from source |
| 06:05.59 | Supaplex | or make your own spec =) |
| 06:06.34 | Derkommissar | :( |
| 06:06.46 | Derkommissar | so no1 here has asterisk working with r2, |
| 06:06.54 | Derkommissar | or gotten chan_unicall to work ? |
| 06:11.56 | linsys | If I have an extension configured in sip.conf and extensions.conf and there is no phone on the extension but I set it up like this in extensions.conf |
| 06:11.56 | linsys | exten => 2001,1,Dial(SIP/2001,20) |
| 06:11.56 | linsys | exten => 2001,2,Voicemail(u2001) |
| 06:11.56 | linsys | exten => 2001,102,Voicemail(b2001) |
| 06:11.56 | linsys | exten => 2001,103,Hangup |
| 06:12.11 | linsys | and I try and dial that extension from lets say extension 2000 I should go to vmail? |
| 06:12.35 | *** part/#asterisk TivoTechie (~info@ool-44c4d842.dyn.optonline.net) |
| 06:13.52 | Derkommissar | not folowing your question |
| 06:14.04 | Derkommissar | you dont set up "extentions" in sip.conf |
| 06:14.36 | linsys | you don't? |
| 06:14.41 | linsys | this is what I have in my sip.conf |
| 06:14.41 | linsys | [2000] |
| 06:14.41 | linsys | type=friend ; This device takes and makes calls |
| 06:14.41 | linsys | username=2000 ; Username on device |
| 06:14.41 | linsys | secret=password ; Password for device |
| 06:14.41 | linsys | host=dynamic ; This host is not on the same IP addr every time |
| 06:14.42 | nine76 | If anyone has played with areskicc I'd appreciate them telling me how to get past the last steps from wiki... i.e. http://pastebin.ca/8710 please,and thx:-/ |
| 06:14.43 | linsys | mailbox=100 ; Activate the message waiting light if this |
| 06:15.02 | linsys | sorry maybe that isn't an extension? |
| 06:15.07 | iq | linsys, pastebin.com |
| 06:15.42 | linsys | ?? |
| 06:15.44 | linsys | iq?? |
| 06:16.06 | iq | linsys, its good to use pastebin.com to paste lots of info |
| 06:16.35 | linsys | oh... |
| 06:17.18 | linsys | well basicly I don't have 2999 setup in sip.conf because I just want to use it to check vmail.. but when I dial 2999 I just get a fast busy after like 20 seconds.. |
| 06:17.25 | linsys | and this error |
| 06:17.26 | linsys | Apr 2 03:10:53 WARNING[15781]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call DB763D4F-84F2-4258-AB75-197EEA3E4858@192.168.0.100 for seqno 48736 (Non-critical Response) |
| 06:21.45 | linsys | oh shit |
| 06:21.46 | linsys | I got it |
| 06:21.49 | linsys | ;) |
| 06:21.52 | Derkommissar | wierd |
| 06:22.00 | Derkommissar | you dont have to set that up in sip.conf |
| 06:22.02 | linsys | I had the listen port set wrong in X-Free.. |
| 06:22.07 | linsys | oh really? |
| 06:22.12 | linsys | that's what this article said |
| 06:22.14 | linsys | so.. ?? |
| 06:22.15 | linsys | I tried it |
| 06:22.19 | linsys | I can now dial 2999 |
| 06:22.26 | linsys | and get my vmail answering service |
| 06:22.32 | linsys | or check vmail service |
| 06:22.35 | Derkommissar | good :-) |
| 06:22.52 | *** part/#asterisk mkhan (~mkhan@dsl092-066-137.bos1.dsl.speakeasy.net) |
| 06:22.57 | Derkommissar | im glad to know people like SIP more than any other protocoll |
| 06:23.10 | linsys | is that bad? |
| 06:23.20 | linsys | I just read some article and tried to follow it... |
| 06:23.28 | Derkommissar | no is not |
| 06:23.30 | Derkommissar | is good |
| 06:23.32 | Derkommissar | :-) |
| 06:23.54 | Derkommissar | so what is it that you are trying to acomplish ? |
| 06:25.17 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 06:25.53 | linsys | eventually I want to get my asterisk box to connect to an analogue line with an X100P |
| 06:26.00 | linsys | but I wanted to get 2 extensions to work first |
| 06:26.06 | linsys | just to see if I had half a clue |
| 06:26.07 | linsys | ha |
| 06:26.08 | _SMP_ | Hi folks, I'm having some trouble updating the firmware on my Cisco 7960 from 6.3 to 7.x |
| 06:26.23 | _SMP_ | Does anyone have any experience with that? |
| 06:26.27 | Derkommissar | cool :-) |
| 06:26.33 | Derkommissar | DAM IT |
| 06:27.00 | Derkommissar | has anyone got channels_makefile.patch for chan_unicall to work ever ? |
| 06:27.56 | Derkommissar | OT ? |
| 06:28.04 | _SMP_ | off topic |
| 06:28.24 | _SMP_ | I spent almost all day upgrading that thing from SCCP to SIP 2.x -> 6.3 |
| 06:28.35 | _SMP_ | Now it's the last push and I can't seem to get 7.x on there |
| 06:29.20 | Derkommissar | :-/ |
| 06:29.25 | *** join/#asterisk moy (~moy@201.138.195.20) |
| 06:29.31 | Derkommissar | sorry, never had the privilege of using one of those |
| 06:29.40 | Derkommissar | i only use cheap phones like grandstream |
| 06:30.06 | _SMP_ | I've been using a SPA-3k until now. Worked like a charm |
| 06:30.22 | _SMP_ | But since I have this hella expensive thing, might as well configure it. |
| 06:30.40 | Jim^^ | _SMP_: what's it doing, just acting like there's no update? |
| 06:33.46 | *** join/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net) |
| 06:33.47 | linsys | Also I see this message in my logs |
| 06:33.48 | linsys | Unable to open /dev/dsp: No such file or directory |
| 06:34.02 | Derkommissar | your sound card is not configured |
| 06:34.02 | linsys | or I ment this |
| 06:34.03 | linsys | Unable to open pseudo channel for timing... Sound may be choppy. |
| 06:34.10 | Derkommissar | your sound card is not configured |
| 06:34.16 | linsys | I'm not sure there is a sound card on this box.. |
| 06:34.22 | linsys | that's what I thought that error was.. |
| 06:34.24 | Derkommissar | and you dont have a digium card |
| 06:34.39 | gtigene | Is insomnia a necessary consequence of having a new *? |
| 06:34.52 | linsys | do I need a good sound card in the asterisk box? |
| 06:34.57 | Derkommissar | Nah |
| 06:35.03 | Derkommissar | just compile ztdummy |
| 06:35.21 | linsys | I do have a Digium card, that's what I'm about to configure now.. |
| 06:35.27 | Derkommissar | gtigene, Its a sign that something is not rigth |
| 06:35.45 | Derkommissar | linsys, then you dont need ztdummy |
| 06:36.00 | gtigene | Derkommissar, I learned about disaster recovery this weekend |
| 06:36.19 | linsys | I have a Wildcard TDM400P |
| 06:36.25 | linsys | with one FXO port on it |
| 06:36.40 | Derkommissar | gtigene, Im learning that r2 and asterisk dont get along good. |
| 06:36.55 | Derkommissar | And i need to find someone to help me out |
| 06:37.32 | gtigene | Derkomissar, r2 would be, uh, rest-and-recreation? |
| 06:37.33 | Derkommissar | its harsh when you cant even compile something yourselft |
| 06:37.54 | Derkommissar | [root@prueba channels]# patch <channels_makefile.patch.1 patching file Makefile |
| 06:37.54 | Derkommissar | Hunk #1 FAILED at 72. |
| 06:37.54 | Derkommissar | Hunk #2 FAILED at 143. |
| 06:37.54 | Derkommissar | Hunk #3 FAILED at 178. |
| 06:37.54 | Derkommissar | 3 out of 3 hunks FAILED -- saving rejects to file Makefile.rej |
| 06:38.20 | Derkommissar | gtigene, No r2, as in you will burn in hell r2. |
| 06:38.48 | Derkommissar | :-( |
| 06:40.06 | tzafrir | Derkommissar, what patch is that? |
| 06:40.20 | Derkommissar | for chan_unicall |
| 06:40.56 | Derkommissar | i gone back and forth all over the cvs, trying to find a vertion where it would maybe work |
| 06:40.59 | tzafrir | Derkommissar, patches to the makefile basically add a number of lines. Try applying them manually |
| 06:41.01 | Derkommissar | But no luck :-( |
| 06:41.07 | Derkommissar | I did |
| 06:41.17 | gtigene | Derkommissar, maybe I am dense but I don't get it. I hope your patch works soon, though |
| 06:41.18 | tzafrir | Just be aware of converting tabs to spaces if you simply copy&paste text. |
| 06:41.36 | Derkommissar | then it shows a bunch of messages |
| 06:41.49 | tzafrir | applying them manually, as in: not using patch |
| 06:42.12 | tzafrir | what's unicall good for, btw? |
| 06:42.46 | Derkommissar | R2 |
| 06:43.07 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 06:45.35 | Derkommissar | afther i apply the patch manually |
| 06:45.37 | Derkommissar | i get this |
| 06:45.38 | Derkommissar | chan_modem.c:1044: error: too many arguments to function `ast_channel_register' |
| 06:45.38 | Derkommissar | make: *** [chan_modem.o] Error 1 |
| 06:46.53 | *** join/#asterisk Inv_arp (junya@adsl-3-255-42.mia.bellsouth.net) |
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| 07:04.34 | *** join/#asterisk stdio (~stdio@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 07:04.43 | linsys | so can anyone tell me if I get a VoIP provider and use a high speed data connection like a 1.5mbps connection can I have a VoIP provider give me a number and setup like 10 extensions using that same number and have all 10 lines make inbound and outbound calls? |
| 07:05.04 | linsys | or should I say have the 10 extensions make 10 inbound and/or outbound calls? |
| 07:05.42 | linsys | This would probably have to be like a true VoIP provider who can give me a SIP termination for my asterisk box, and not someone like Vonage or what ever.. right? |
| 07:05.56 | linsys | 10 concurrent connections that's what I'm looking for.. |
| 07:06.16 | stdio | hey all.... |
| 07:08.56 | stdio | so, let's say extension 123 is ringing... and I want to pull it over to my extension (125), and take the call (probably because the person at 123 isn't there, and I want to take their call so it doesn't go to voicemail.).... a) what is this "feature" called b) is the answer to this question phone specific? (spa-841) c) anyone know how to do it? |
| 07:09.26 | *** join/#asterisk rious (~rious@adsl-67-36-57-236.dsl.klmzmi.ameritech.net) |
| 07:10.24 | rious | has anybody used the rxfax and txfax apps ? |
| 07:11.10 | Derkommissar | sdp |
| 07:11.22 | Derkommissar | visit soft-switch.org |
| 07:11.30 | *** join/#asterisk jdiskywlkr (~kvirc@ip68-0-90-1.tu.ok.cox.net) |
| 07:11.32 | Derkommissar | there is a good one there |
| 07:11.39 | rious | I've got them working |
| 07:12.12 | rious | I can recieve faxes great via sip, but I have alot of trouble using IAX, is there a jitter/quality difference between the two ? |
| 07:12.21 | Derkommissar | then whats the problem ? |
| 07:12.45 | *** join/#asterisk shodan (~shodan@216.113.99.166) |
| 07:14.59 | gtigene | Yawn |
| 07:15.34 | gtigene | Thanks everybody. Good night :) |
| 07:15.41 | *** part/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net) |
| 07:31.06 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
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| 08:42.40 | *** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net) |
| 08:49.30 | *** join/#asterisk |Vulture| (~Vulture@152.238.204.68.cfl.res.rr.com) |
| 08:49.42 | |Vulture| | Anyone use QoS, DiffServ on a Netgear switch? |
| 08:50.58 | *** join/#asterisk Warp[mh] (~warp@217.106.87.250) |
| 08:55.01 | patdk | yep |
| 08:55.11 | patdk | you need to upgrade to latest firmare |
| 08:55.25 | patdk | cause it doesn't let you set enough flags on the other ones |
| 08:57.11 | |Vulture| | oh damn I got like 6 in the field they are all 2.0.0 |
| 08:57.17 | |Vulture| | they are on like 2.6.0 now lol |
| 08:57.34 | |Vulture| | patdk: what do I set? I am looking at the DiffServ Priority: |
| 08:57.50 | |Vulture| | and see 63 fields with normal/high |
| 08:58.42 | patdk | hmm, I was using like 3.0.2 |
| 08:59.03 | |Vulture| | Ive got a FSM726] |
| 08:59.15 | |Vulture| | I am sure the user interface is pretty close |
| 08:59.17 | |Vulture| | its a Layer2 |
| 08:59.19 | patdk | hmm, don't know about that |
| 08:59.31 | patdk | I use the fsm7326 and gsm7312 |
| 09:00.06 | |Vulture| | what phones do you use with the poe? |
| 09:00.12 | patdk | I never saw a normal/high on min |
| 09:00.20 | patdk | snom |
| 09:00.38 | darkskiez | woo glastonbury tickets |
| 09:00.50 | |Vulture| | patdk: what did you have to set? |
| 09:01.04 | patdk | set? |
| 09:01.04 | |Vulture| | Ill look at the fsm7326 manual and see if it is similar |
| 09:01.13 | |Vulture| | for QoS to work with VoIP |
| 09:01.24 | patdk | you just use diffserv |
| 09:01.28 | patdk | you mark incoming packets |
| 09:01.43 | patdk | and you prioritorize outgoing packets |
| 09:02.08 | patdk | heh, it even has an auto voip qos setup on the firmware |
| 09:02.21 | patdk | if you want simple, and that is all you want to do with diffserv |
| 09:03.01 | |Vulture| | hmm strange |
| 09:03.28 | |Vulture| | trying to find a pic of the web interface see if it is similar |
| 09:05.10 | *** join/#asterisk eye69 (magnus@upcore.net) |
| 09:05.33 | |Vulture| | ah I see you have a Diffserv Wizard |
| 09:08.24 | *** join/#asterisk shodan (~shodan@216.113.99.174) |
| 09:09.43 | patdk | damn perl |
| 09:09.54 | patdk | I can't figure out how to access these variables |
| 09:11.29 | dieck | $, @, % :) |
| 09:11.47 | patdk | no, scope, not type |
| 09:11.59 | dieck | was made so that cursing irc users can easily remember :) |
| 09:12.06 | dieck | ah, scope, ok, that's another problem |
| 09:12.30 | patdk | main defines some vars |
| 09:12.35 | patdk | I need a sub to access them |
| 09:12.57 | dieck | just give a reference to the sub? |
| 09:13.11 | patdk | reference as in pass? |
| 09:13.30 | dieck | reference as in subname(variables used) |
| 09:13.48 | patdk | no, can't do it that way |
| 09:14.11 | dieck | hm |
| 09:14.14 | harryvv | I opened up rtp ports 10001-10012 on my end and the other end user opened up 10001 would that still cause voice problems? |
| 09:14.23 | dieck | i used to program a lot in perl, but i switched to php some years ago |
| 09:14.26 | patdk | harryvv, probably |
| 09:14.32 | patdk | dieck, heh, I don't use perl |
| 09:14.35 | dieck | there it's "global $var;" |
| 09:14.43 | dieck | but I think perl was different |
| 09:14.46 | harryvv | pat, do you have rtp opened up for ports outside your network? |
| 09:14.47 | patdk | but this one program is init, and well, don't want to rewrite it |
| 09:15.00 | patdk | harryvv, ya, 10000-20000 |
| 09:15.30 | harryvv | Im stuck with a simple soho router that does not have that wide range of ports to put into it. |
| 09:15.44 | patdk | than you have to limit asterisk |
| 09:15.59 | patdk | remember, every voice connection will use 4 rtp ports |
| 09:16.03 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 09:16.05 | patdk | or was it two, heh |
| 09:16.25 | implicit | anyone know where to get firmware for the hitachi phone? |
| 09:16.39 | implicit | wip-5000 |
| 09:16.56 | harryvv | pat, which 4 rtp ports? |
| 09:17.18 | patdk | harryvv, any you define |
| 09:17.31 | harryvv | so it needs a min of 4 for voice to go both ways? |
| 09:17.51 | patdk | I think just two, but 4 is safe |
| 09:18.05 | dieck | patdk: looks like perl gives scope to subroutines automatically. just try $var in the sub |
| 09:18.27 | patdk | dieck, doesn't work, gives unknown var error |
| 09:18.39 | patdk | I tried $main::var, but that just gves me a blank var |
| 09:18.44 | dieck | argh |
| 09:18.51 | dieck | just say you work object orientated |
| 09:19.15 | dieck | it's different there |
| 09:19.19 | patdk | heh, I have no idea, it's how ever this people did it |
| 09:19.34 | patdk | heh,I tried $var,no good, gives compile error |
| 09:19.46 | dieck | therefore I have to look into the Panther, not into the Camel |
| 09:20.37 | patdk | hmm, I think I got it :) |
| 09:20.41 | harryvv | <PROTECTED> |
| 09:20.47 | harryvv | still no luck |
| 09:20.48 | patdk | using $main::var and defining it as, our $var |
| 09:20.52 | *** join/#asterisk tessier (~treed@210.245.38.7) |
| 09:21.10 | patdk | harry, you did edit rtp.conf? |
| 09:21.23 | harryvv | yes and need to look at it again. |
| 09:22.11 | harryvv | ; |
| 09:22.11 | harryvv | ; RTP start and RTP end configure start and end addresses |
| 09:22.12 | harryvv | ; |
| 09:22.12 | harryvv | rtpstart=10000 |
| 09:22.12 | harryvv | rtpend=10010 |
| 09:22.14 | harryvv | ; |
| 09:22.16 | harryvv | ; Whether to enable or disable UDP checksums on RTP traffic |
| 09:22.22 | harryvv | ; |
| 09:22.24 | harryvv | ;rtpchecksums=no |
| 09:22.36 | harryvv | he has 10001-10004 open |
| 09:22.56 | harryvv | pat you here? |
| 09:23.05 | *** join/#asterisk spongie (~ob1@70-57-11-107.dnvr.qwest.net) |
| 09:23.14 | patdk | you forwarding 5060? |
| 09:23.19 | harryvv | yes |
| 09:23.35 | patdk | your behind a firewall, and so is he? |
| 09:23.43 | harryvv | I can see him trying to access the voicemail but he does not hear anything. |
| 09:23.43 | |Vulture| | is * default DSCP 28? |
| 09:25.13 | harryvv | yes routers |
| 09:29.35 | ta[i]nted | that quit message just doesn't read quite right |
| 09:31.52 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 09:36.46 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 09:38.17 | *** join/#asterisk Zgarbi (Kirile@ppp119-cs1.sanet.ge) |
| 09:38.31 | Zgarbi | hi |
| 09:40.31 | Zgarbi | while compile libpri from just updated cvs it gives me error: |
| 09:40.32 | Zgarbi | q931.c: In function 'send_message': |
| 09:40.33 | Zgarbi | q931.c:2503: warning: pointer targets in passing argument 3 of 'init_header' differ in signedness |
| 09:41.53 | Zgarbi | any solution? |
| 09:42.55 | *** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 09:43.14 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 09:45.44 | lesouvage | <PROTECTED> |
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| 09:46.41 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
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| 09:55.01 | spongie | <PROTECTED> |
| 09:55.06 | spongie | and suggestions? |
| 09:55.11 | spongie | any* |
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| 10:12.24 | WilliamK | can anyone verify for me the line in zaptel.conf to use for clocking off the telco's switch (clock source line) |
| 10:13.25 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-207-192.dsl.scarlet.be) |
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| 10:32.29 | *** join/#asterisk gburger (~g@myw-stp-196-37-192-64.sentechsa.net) |
| 10:33.04 | gburger | can asterisk proxy h323 connections? |
| 10:33.49 | cypromis | no |
| 10:33.56 | cypromis | you need a gatekeeper for that |
| 10:34.44 | gburger | ok, any suggestions? |
| 10:34.50 | gburger | linux/gentoo |
| 10:36.19 | gburger | gnugk? |
| 10:37.26 | cjk | hi, i have coded a click to dial funtion in my webpage which does the socket connection asterisk-manager. but in that case the sip-rtp stream does always pass the *. canreinvite is ignored. any idea why? will it be the same with iax2? |
| 10:37.37 | dieck | gburger: http://www.voip-info.org/tiki-index.php?page=Asterisk%20H323%20channels |
| 10:40.06 | Nix | cjk: iax does not have the capability of a separate audio stream.. |
| 10:40.11 | gburger | k, so if a person is behind a cable router that does NAT, then a h323 connection would be pretty impossible |
| 10:40.32 | cjk | Nix, so? will the traffic pass through * or not? |
| 10:40.37 | spongie | <PROTECTED> |
| 10:40.44 | spongie | any suggestions? |
| 10:41.08 | Nix | cjk: where are you iniating the dial from? the * box? |
| 10:41.30 | cjk | Nix, well yeah i to an originate-call in astman or a .call file |
| 10:41.37 | cjk | Nix, well yeah i do an originate-call in astman or a .call file |
| 10:42.14 | Nix | so the audio is comming from * |
| 10:42.19 | Nix | what is the problem exactly? |
| 10:42.24 | *** join/#asterisk h3x0r (Justino@ip70-180-167-6.lv.lv.cox.net) |
| 10:42.28 | h3x0r | haha |
| 10:42.29 | cjk | i do not want the audio go through * |
| 10:42.32 | Nix | are you connecting that call to 2 different endpoints? |
| 10:42.34 | Nix | ahhh |
| 10:42.36 | h3x0r | voip-info.org is hosted like 50 feet away from my datacenter |
| 10:42.51 | Nix | are you using sip on both ends of the call? |
| 10:42.57 | cjk | Nix, yeah i connect sip user 1 with sip user 2 |
| 10:42.59 | Nix | * in the middle |
| 10:43.02 | Nix | ok |
| 10:43.11 | cjk | Nix, both are registered on * |
| 10:43.27 | cjk | * should be in the middle for the sip traffic, but the rtp traffic normally does not pass * |
| 10:43.29 | Nix | it should be possible |
| 10:43.32 | cjk | if canreinvite is set to yes |
| 10:43.53 | cjk | it work when i dial from phone, but when * originates the call to make the link it does not work |
| 10:44.14 | Nix | I think you actually need to connect the call, and then transfer it for the reinvite to work |
| 10:44.31 | cjk | ok |
| 10:44.36 | cjk | i will try it out, tahnks |
| 10:44.39 | Nix | gburger: you definately need gnugk as cypromis says |
| 10:47.58 | gburger | what protocol/program is the best for video+voice thru NAT'ed firewalls? |
| 10:51.32 | h3x0r | video? |
| 10:51.35 | h3x0r | you are screwed |
| 10:51.35 | h3x0r | haha |
| 10:52.30 | cypromis | which OS ? |
| 10:53.00 | cypromis | the new gnomemeeting will be OPAL based so will support at least h.323 and sip |
| 10:53.06 | Nix | gburger: a coax cable is generally the best ;-) |
| 10:53.17 | cypromis | dunno about windows although the xten eyebeamer stuff or how it's called should probably even work |
| 10:53.20 | h3x0r | hahahaha |
| 10:53.43 | h3x0r | coax |
| 10:53.53 | Nix | or fibre ;-) |
| 10:54.04 | cypromis | coax is ok |
| 10:54.12 | cypromis | as long as it end in E3 interfaces on both ends |
| 10:54.24 | Nix | yep. lol |
| 10:54.32 | Nix | bah. |
| 10:54.33 | cypromis | :) |
| 10:54.47 | Nix | we just rented a new office |
| 10:54.52 | Nix | and applied for new phone lines |
| 10:55.00 | cypromis | hehe |
| 10:55.03 | cypromis | sounds familiar |
| 10:55.06 | Nix | signed the contract and were given our new phone numbers |
| 10:55.32 | Nix | so we got new cards printed thursday morning for the telekom2005.com fair thursay-Saturday |
| 10:55.49 | Nix | thursday evening they connected the phones.. with different numbers.. |
| 10:55.50 | Nix | fuckers |
| 10:56.25 | spongie | <PROTECTED> |
| 10:56.27 | spongie | any suggestions? |
| 10:56.54 | cypromis | did you edited zaptel.conf for it ? |
| 10:57.03 | spongie | I can see it sees the card because I get some message in /var/log/messages voip kernel: Zapata Telephony Interface Registered on major 196 |
| 10:57.07 | spongie | but that's all it says |
| 10:57.14 | spongie | yes I added this into the zaptel.conf |
| 10:57.30 | spongie | loadzone=us |
| 10:57.30 | spongie | defaultzone=us |
| 10:57.30 | spongie | fxsks=1 |
| 10:58.07 | spongie | then I run ztconfig |
| 10:58.09 | spongie | and i get this |
| 10:58.12 | spongie | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 10:58.12 | spongie | 1 channels configured. |
| 10:58.12 | spongie | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 10:58.20 | spongie | I also see the dir |
| 10:58.21 | spongie | ops |
| 10:58.37 | gburger | cypromis: but with gnomemeeting i will still require a gatekeeper and so will the other person if we are both behind NAT firewalls? |
| 10:59.10 | spongie | I also see the dir created /dev/zap/channel; ctl; pseudo; and timer, however I shoudl also see a 1 or 2 or 3 or 4 in my case should be a number 1 |
| 10:59.11 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 10:59.18 | spongie | this system uses udev |
| 10:59.29 | *** join/#asterisk eivindtr (~eivind@062016241059.customer.alfanett.no) |
| 10:59.53 | spongie | so I also added in the enteries in the /etc/udev/permissions.d/50-udev.permissions |
| 11:00.08 | spongie | and in ../rules.d/50-udev.rules |
| 11:00.11 | cypromis | gburger: when using SIP you will require a sip proxy |
| 11:00.16 | cypromis | or a server on an outside ip |
| 11:00.22 | cypromis | with h.323 you will require a gatekeeper |
| 11:00.27 | spongie | I added info from the README.udev... anyone had this issue before? |
| 11:00.52 | cypromis | did you try to put the card in a different pci slot ? |
| 11:00.55 | spongie | yes |
| 11:00.59 | spongie | 3 different ones.. . |
| 11:01.10 | spongie | there was a pci nic card in one of the slots before and it worked fine.. |
| 11:01.18 | spongie | maybe it's the box |
| 11:01.23 | spongie | are there certain requirements? |
| 11:01.36 | spongie | I have found some and even read of people who where running this on some PII I think.. |
| 11:01.46 | spongie | this is a PII450 with 128MB ram... |
| 11:02.24 | gburger | cypromis: can one do video over sip? soz for all the question, just give me url thats good |
| 11:04.11 | cypromis | http://www.xten.com/index.php?menu=products&smenu=eyebeam |
| 11:04.14 | cypromis | is a good example |
| 11:06.11 | *** join/#asterisk [shodan] (~shodan@216.113.99.180) |
| 11:07.46 | gburger | it seems like sip suffers from the same problems as h323 over nat firewalls, so that wont work either, IAX seems like a alternative |
| 11:08.56 | cypromis | yeah but I haven't seen any video client for iax yet |
| 11:10.31 | gburger | so if i have a cable router that does basic nat/dnat i pretty much screwed |
| 11:14.50 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
| 11:15.05 | dwmw2_gone | just roll out ipv6 and stop playing with nat :) |
| 11:15.22 | dwmw2_gone | it's not as if ipv6 is hard |
| 11:15.56 | gburger | over current inet you have to v6 over v4 which is a waste imho |
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| 11:24.39 | Vhaway | anyong using voipjet for Intl? |
| 11:25.00 | Vhaway | I can't seem to get voipjet to do Intl, work fine for US |
| 11:25.15 | *** join/#asterisk rndmtngnt ([U2FsdGVkX@203.13.74.236) |
| 11:26.02 | Vhaway | no one here? |
| 11:27.29 | *** join/#asterisk IP_Fox (~barny@cpc1-char1-5-0-cust71.sot3.cable.ntl.com) |
| 11:29.15 | Vhaway | oh/qui |
| 11:30.40 | IP_Fox | can anyone help me with a real weird * to * problem using IAX? |
| 11:35.28 | Supaplex | I dunno |
| 11:35.33 | Supaplex | Can they? |
| 11:35.54 | Supaplex | <<-- smart arse |
| 11:37.09 | Supaplex | assume we can, just ask a smart(er) question, and if someone can help, they will (rule#1 ask an open ended question often) |
| 11:37.20 | IP_Fox | okay |
| 11:37.24 | IP_Fox | so... |
| 11:37.52 | IP_Fox | IAX2 between to * servers. Bi directional trunk. One * server is behind NAT, other is public IP in telehouse |
| 11:38.10 | Supaplex | k. define weird problem |
| 11:38.23 | IP_Fox | the * in telehouse uses SIP to interface with SIPGATE, FWD etc etc, and forwards all calls via IAX2 to the NAtted * |
| 11:38.29 | IP_Fox | incoming calls work fine |
| 11:38.50 | IP_Fox | when outgoing calls are initiated, the natted * box wont dial with IAX |
| 11:38.51 | h3x0r | my first guess is codec negotiation |
| 11:39.11 | IP_Fox | but if i dial inbound, then wait a few secs and make an outbound call everything works |
| 11:40.27 | h3x0r | what direction is inbound |
| 11:40.27 | IP_Fox | so its like the * natted box cant get through to the public * box unless the public * box has called it moments before |
| 11:40.27 | IP_Fox | inbound is telehouse -> NAT |
| 11:40.27 | h3x0r | hm |
| 11:40.27 | h3x0r | what codec are you using |
| 11:40.30 | h3x0r | and are you using asterisk behind nat or is it an iaxy or something |
| 11:40.44 | IP_Fox | i am getting it to use any codec avail |
| 11:40.57 | h3x0r | what codecs do your sip providers have |
| 11:40.58 | IP_Fox | behind NAT with PAT on the IAX port |
| 11:41.07 | IP_Fox | 711 |
| 11:41.29 | IP_Fox | but the thing is, it does work... |
| 11:41.31 | h3x0r | hmmm. |
| 11:41.42 | IP_Fox | but only if it has made a connection attempt inbound moments before |
| 11:42.01 | h3x0r | usually a problem like that would be reverse |
| 11:42.05 | h3x0r | because of a nat mapping timeout |
| 11:42.06 | IP_Fox | when i debug IAX, i see the natted * box calling the public * box, but the public box doesnt responbd |
| 11:42.10 | IP_Fox | sure |
| 11:42.17 | h3x0r | do you have nat=yes on both ends |
| 11:42.19 | h3x0r | or one side |
| 11:42.29 | IP_Fox | hold on... |
| 11:42.33 | IP_Fox | i just check ;) |
| 11:42.34 | h3x0r | in iax.conf |
| 11:43.17 | h3x0r | er nevermind that only applies to sip.conf |
| 11:43.20 | h3x0r | what am i thinking |
| 11:43.28 | IP_Fox | i just got it on the natted box |
| 11:43.40 | IP_Fox | i assumed you mean that! |
| 11:44.02 | h3x0r | whats your dial line look like in extensions |
| 11:44.03 | IP_Fox | and i have done a netstat on the public box, and the ports are listening |
| 11:44.13 | h3x0r | are you using @nameofcontextiniaxconf |
| 11:44.21 | h3x0r | or did you put an ip address or dns there |
| 11:44.30 | IP_Fox | exten => _902.,1,Dial(IAX2/exeye:XXXXX@83.170.75.85/${EXTEN}) |
| 11:44.36 | IP_Fox | ip |
| 11:44.42 | h3x0r | well thats what your problem is |
| 11:44.43 | h3x0r | i think |
| 11:44.56 | IP_Fox | i should use FQ DNS |
| 11:44.57 | h3x0r | actually its IAX2/profilename/number |
| 11:44.59 | h3x0r | no |
| 11:45.06 | h3x0r | profilename as in iax.conf |
| 11:45.20 | h3x0r | finally, do you have trunking turned on |
| 11:45.34 | IP_Fox | in iax.conf? |
| 11:46.13 | h3x0r | you shouldnt have trunking enabled unless you have a zaptel timing device on both ends |
| 11:46.26 | h3x0r | trunk=yes |
| 11:46.26 | IP_Fox | no thats definately turned off |
| 11:46.30 | h3x0r | ok |
| 11:46.38 | IP_Fox | no zaptel, completely IP |
| 11:47.12 | Supaplex | IP_Fox: nothing personal there. sorry if I'm a little harsh. I wanted to be in bed a few hours ago, and I tend to be more outspoken when I'm grumpy. |
| 11:47.14 | h3x0r | anyway |
| 11:47.25 | h3x0r | as i was saying |
| 11:47.26 | Supaplex | take care. I'm off to Zzz land. |
| 11:47.33 | IP_Fox | bye ;o) |
| 11:47.35 | DrukenHME | h3x0r: you should have a zaptel timing device on every machine anyways |
| 11:47.53 | h3x0r | you have a configuration in iax.conf for your provider like say [fwd] |
| 11:48.00 | h3x0r | drunk: It does nothing except for meetme and iax2 trunking |
| 11:48.06 | IP_Fox | in my sip.conf, yes |
| 11:48.21 | h3x0r | no other application potentially touches it in voip land |
| 11:48.27 | DrukenHME | h3x0r: really.. try doing musiconhold without one :) |
| 11:48.28 | IP_Fox | nope |
| 11:48.33 | h3x0r | er sorry i mean |
| 11:48.36 | h3x0r | your iax stuff |
| 11:48.38 | h3x0r | like |
| 11:48.47 | IP_Fox | so, i dont need to specify secrets in the dial string then? |
| 11:48.54 | h3x0r | you have a [telehouse] in iax.conf on your natted box right |
| 11:49.01 | IP_Fox | yeah |
| 11:49.04 | IP_Fox | called [exeye] |
| 11:49.11 | h3x0r | so your extensions.conf dial line should be IAX2/telehouse/${EXTEN}..... |
| 11:49.15 | h3x0r | Ok |
| 11:49.17 | h3x0r | yeah |
| 11:49.27 | IP_Fox | right, think thats the problem then |
| 11:49.30 | h3x0r | then its IAX2/exeye/${EXTEN} |
| 11:49.39 | IP_Fox | weird how it works sometimes though, y would that b? |
| 11:49.59 | h3x0r | because it still has a connection established between the two peers |
| 11:50.05 | h3x0r | and uses the cached settings i guess |
| 11:50.16 | IP_Fox | right, |
| 11:50.30 | IP_Fox | info on the WIKI is hard to find on this particular subj... |
| 11:50.31 | h3x0r | it definately uses the settings in iax.conf on inbound calls |
| 11:50.38 | h3x0r | because thats how it figures out what context to start running |
| 11:50.51 | h3x0r | well, the way its all set up is terrible :) |
| 11:50.57 | IP_Fox | guess it wasnt so weird after all!!! |
| 11:51.23 | IP_Fox | thank you, you have saved me loads of grey hairs!!! |
| 11:51.30 | h3x0r | heh |
| 11:51.45 | h3x0r | one thing that ive noticed asterisk totally sucks at is passing sip -> sip |
| 11:51.48 | h3x0r | i cant get the shit to work right |
| 11:52.00 | h3x0r | it just totally fubars codec negotiation |
| 11:52.03 | *** join/#asterisk L|NUX (linux@202.5.129.98) |
| 11:52.08 | h3x0r | im switching my core to SER coz its pissing me off |
| 11:52.31 | IP_Fox | i have such a complex setup to overcome NAT issues with SIP... |
| 11:52.56 | h3x0r | well its sorta easy |
| 11:52.57 | IP_Fox | coz i have client with Cisco Call Manager wanting to use SIP trunks to things like FWD etc etc |
| 11:53.00 | h3x0r | if you set up a stun proxy or something |
| 11:53.05 | IP_Fox | and CCM only allows 1 SIP trunk.. |
| 11:53.10 | DrukenHME | h3x0r: it does? i've had my stuff pass sip to sip no problem... |
| 11:53.13 | h3x0r | ccm is the devil |
| 11:53.21 | h3x0r | Druken: it works fine if its g.711 to g.711 |
| 11:53.22 | IP_Fox | tell me about it... |
| 11:53.30 | h3x0r | but i cant get it to do things like g.729 to g.729 |
| 11:53.36 | h3x0r | or g.723.1 to g.723.1 |
| 11:53.36 | DrukenHME | g.729a.... |
| 11:53.39 | h3x0r | yeah a |
| 11:55.31 | IP_Fox | k so i just made those changes, my dial line looks like | exten => _901.,1,Dial(IAX2/exeye/${EXTEN}) |
| 11:55.36 | IP_Fox | still no luck... |
| 11:56.29 | h3x0r | iax2 debug |
| 11:56.44 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 11:57.25 | IP_Fox | just reading now.. |
| 11:58.02 | *** join/#asterisk Pantanero (~Pantanero@195-23-244-106.net.novis.pt) |
| 11:58.06 | IP_Fox | well its dailling the right address... |
| 11:59.13 | IP_Fox | again, on the public * box, iax debug shows nothing, as if its receiving nothing from the natted * box... |
| 11:59.38 | h3x0r | that is strange |
| 11:59.43 | h3x0r | maybe your firewall sucks ass |
| 11:59.52 | IP_Fox | i have turned all firewalls off |
| 12:00.02 | h3x0r | you didnt tell it to forward a port inbound did you |
| 12:00.15 | h3x0r | 4569? |
| 12:00.28 | h3x0r | Oh I know what the problem is maybe |
| 12:00.31 | IP_Fox | on that NAT i did |
| 12:00.36 | h3x0r | you need to do a register line |
| 12:01.06 | h3x0r | well i guess that would affect the inbound more than outbound. hm |
| 12:01.06 | DrukenHME | i feel i must say that in my own opinion, a voip server should NEVER be behind a NAT |
| 12:01.13 | h3x0r | it isnt |
| 12:01.16 | *** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
| 12:01.24 | h3x0r | thats his customer end |
| 12:01.30 | DrukenHME | oh... |
| 12:01.39 | DrukenHME | well then, that's a diffrent story |
| 12:01.43 | IP_Fox | ;) |
| 12:01.54 | h3x0r | another thing |
| 12:01.58 | h3x0r | did you do a host= line on both sides |
| 12:02.04 | IP_Fox | yeah |
| 12:02.12 | h3x0r | is it host=dynamic at your telehouse side |
| 12:02.13 | IP_Fox | i paste the iax.conf... |
| 12:02.26 | IP_Fox | no, cos the natted end has static IP... |
| 12:02.41 | IP_Fox | [exeye] |
| 12:02.41 | IP_Fox | type=friend |
| 12:02.41 | IP_Fox | username=exeye |
| 12:02.42 | IP_Fox | secret=xxxx |
| 12:02.42 | IP_Fox | host=81.96.214.71 |
| 12:02.42 | IP_Fox | context=exeyein |
| 12:02.43 | h3x0r | try with dyanmic anyway |
| 12:02.44 | IP_Fox | canreinvite=yes |
| 12:02.50 | jontow | IP_Fox; pastebin! |
| 12:02.54 | jontow | ~pastebin |
| 12:03.02 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 12:03.15 | b0ef | asterisk cvs head is hogging my cpu during iax calls; anyone noticed this? |
| 12:03.16 | IP_Fox | sorry guys... |
| 12:03.19 | jontow | ;) its ok |
| 12:03.25 | jontow | we caught ya before you went nuts |
| 12:03.27 | b0ef | I've never had this problem before |
| 12:04.01 | jontow | b0ef; make sure you check which codecs you're using between hops/servers.. just a thought :/ |
| 12:04.05 | h3x0r | b0ef: its coz im using your box to terminate intelsat calls |
| 12:04.07 | h3x0r | bwahahahahah just kidding |
| 12:04.15 | b0ef | h3x0r: ;) |
| 12:04.22 | h3x0r | $3/Min later... |
| 12:04.31 | b0ef | jontow: no, this is a direct ip2ip call |
| 12:04.48 | b0ef | from asterisk to a sorry ass windows user using iaxcomm |
| 12:05.13 | jontow | hmm.. try it with a different machine / different iax client if possible? |
| 12:05.33 | b0ef | well, they're all using iaxclient |
| 12:05.46 | h3x0r | IP_Fox: Well just for the hell of it host=dynamic and "reload chan_iax2.so" in the CLI |
| 12:05.59 | h3x0r | only on the telehouse side |
| 12:06.09 | h3x0r | and then |
| 12:06.15 | b0ef | I'll do some more tests and post to the ml |
| 12:06.18 | h3x0r | on your nat side, in iax.conf |
| 12:06.20 | IP_Fox | i done a host dynamic, still made no diff, and i did a restart now.. |
| 12:06.30 | h3x0r | register => user:pass@ip |
| 12:08.01 | *** join/#asterisk RoyK (~roy@213.138.231.87) |
| 12:08.12 | RoyK | ~seen coppice |
| 12:08.29 | jbot | coppice <~chatzilla@227.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 6d 3h 47s ago, saying: 'hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest'. |
| 12:08.42 | h3x0r | hahahhahah |
| 12:08.50 | h3x0r | i love last seen messages taken out of context. |
| 12:09.06 | IP_Fox | heh |
| 12:09.35 | jontow | you know whats great.. |
| 12:09.50 | IP_Fox | the register line, appears not to work... iax2 show registry shows state request sent... |
| 12:10.16 | jontow | when you run the phone system, and can go into work on the weekends and just put your phone on speaker, dial an extension and be put into an indefinite music on hold ;) |
| 12:10.23 | h3x0r | blame your stupid nat firewall |
| 12:10.37 | IP_Fox | heh! |
| 12:10.47 | jontow | i have IAX2 working through NAT firewalls |
| 12:10.53 | jontow | .. but all of my NAT firewalls run freebsd :) |
| 12:11.05 | h3x0r | jonas: when i was at paetec colo at one wilshire working on a customer's equipment |
| 12:11.08 | IP_Fox | this is a cisco firewall... |
| 12:11.13 | h3x0r | someone else there had a stupid ass MOH hookup |
| 12:11.16 | jontow | (and potentially are properly configured for the application ..) |
| 12:11.23 | h3x0r | they had a walkman cd player with a cd on repeat all |
| 12:11.26 | h3x0r | plugged into a computer |
| 12:11.27 | IP_Fox | i know its not the firewall, cos i can get the natted * to register to 1899 via iax no probs |
| 12:11.28 | jontow | haha |
| 12:11.30 | h3x0r | next time i go there |
| 12:11.37 | h3x0r | im gonna load a Lords of Acid CD in it |
| 12:11.50 | h3x0r | "show me your pussy, show it to me! let me see your pussy!".... |
| 12:11.53 | jontow | :D |
| 12:11.58 | h3x0r | that would fuckin rule |
| 12:12.08 | jontow | thats a pretty crackheaded way of doing music on hold, these days :( |
| 12:12.12 | h3x0r | yeah well |
| 12:12.19 | h3x0r | they probably had some isa dialogic cards in it too |
| 12:12.26 | jontow | yeah, i found one of those here |
| 12:12.37 | jontow | does * do anything with 'em? |
| 12:12.44 | IP_Fox | can i specify externaddr in iax.conf? |
| 12:12.49 | h3x0r | supposedly but i think only in the commercial * |
| 12:12.53 | jontow | ah :/ |
| 12:12.55 | h3x0r | its implemented crappily |
| 12:13.06 | h3x0r | all it does is nails all channels up to playback/record resources |
| 12:13.09 | jontow | they're the cards that have the DSP on them yes? |
| 12:13.15 | h3x0r | so it acts jsut as dumb as a zaptel device |
| 12:13.45 | h3x0r | IP_Fox: can you run a packet sniffer on your telehouse side |
| 12:14.04 | h3x0r | like tcpdump host = foo or some crap |
| 12:14.08 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 12:14.11 | IP_Fox | 1 min... |
| 12:14.18 | IP_Fox | ill get it going... |
| 12:15.38 | L|NUX | can any one tell me how can i configure MD3200 to make PSTN calls ? |
| 12:15.40 | jontow | question for anybody that may have info.. in queues.conf, if i set joinempty to 'no', and leavewhenempty = 'yes' |
| 12:16.02 | jontow | will that make the Queue() application return any different status to be acted upon? |
| 12:17.09 | Chuji | wow, gmail just increased their storage limit to 2GB |
| 12:17.23 | jontow | that was yesterday ;) now on to conquer the world |
| 12:17.33 | jontow | one free web-based email provider at a time.. potentially two. |
| 12:17.41 | Chuji | You are currently using 426 MB (21%) of your 2053 MB. |
| 12:17.59 | Chuji | 426 MB of Asterisk mail |
| 12:18.03 | Chuji | heh |
| 12:19.22 | h3x0r | gmail would be perfect for warez trading |
| 12:19.22 | h3x0r | haha |
| 12:19.40 | h3x0r | fwd'ing on-net email just like the aol days |
| 12:20.44 | jontow | yeah.. you can access it with POP3 too |
| 12:20.45 | jontow | :P |
| 12:21.59 | RoyK | ~nickometer L|NUX |
| 12:22.11 | RoyK | ~nickometer h3x0r |
| 12:22.17 | h3x0r | oh come on. |
| 12:22.18 | RoyK | :) |
| 12:22.20 | h3x0r | nickometer is stupid |
| 12:22.24 | h3x | fkjhsdafkldjshafdklsjah |
| 12:22.28 | jontow | ~nickometer jontow |
| 12:22.36 | L|NUX | :$ |
| 12:23.00 | Chuji | ~nickometer Chuji |
| 12:23.04 | RoyK | ~nickometer h3x |
| 12:23.07 | Chuji | Had to make sure |
| 12:23.09 | Chuji | :) |
| 12:23.11 | jontow | :D |
| 12:23.19 | GNULinux | ~nickometer GNULinux |
| 12:23.23 | h3x | god damn |
| 12:23.29 | h3x | i just drank some gin for the first time |
| 12:23.39 | h3x | i can drink vodka like a fish but this shit is kicking my ass |
| 12:23.51 | cypromis | lol |
| 12:23.57 | L|NUX | okay :) |
| 12:24.03 | *** join/#asterisk xpasha (~pavel@217.30.252.68) |
| 12:24.04 | cypromis | how much vodka is vodka drunk like a fish ? |
| 12:24.07 | RoyK | ~nickometer RoyK |
| 12:24.15 | h3x | i found a bottle of sapphire bombay in my gf's arsenal |
| 12:24.33 | h3x | cypromis: the way i feel right now it dosent fuckin matter |
| 12:24.34 | h3x | haha |
| 12:24.53 | h3x | pretty soon im gonna look like a qwerty typist trying to use a dvorak keyboard |
| 12:25.55 | cypromis | lol |
| 12:26.09 | IP_Fox | okay, so i solved it! |
| 12:26.17 | RoyK | ~nickometer IP_Fox |
| 12:26.17 | h3x | let me guess |
| 12:26.20 | h3x | you typoed the ip address :P |
| 12:26.26 | IP_Fox | nope. |
| 12:26.27 | RoyK | ~lart h3x |
| 12:26.31 | h3x | man |
| 12:26.47 | h3x | i had a customer coloed with me and last week he couldnt figure out why he couldnt dial local anymore |
| 12:26.49 | RoyK | he mistyped the wrong ip addr |
| 12:26.54 | IP_Fox | heh ;D |
| 12:27.04 | h3x | come to find out he typoed digit deletion of 720 instead of 702 area code |
| 12:27.17 | IP_Fox | i just added a bindaddr=0.0.0.0 to the natted SIP.conf |
| 12:27.23 | h3x | WTF |
| 12:27.33 | IP_Fox | that should have been IAX.conf.. |
| 12:27.45 | h3x | hahahahhahahahaha! |
| 12:27.46 | IP_Fox | and i added nat=yes to the public * iax.conf |
| 12:28.01 | IP_Fox | ;) |
| 12:28.04 | h3x | i dont think nat=yes does anything in iax |
| 12:28.20 | IP_Fox | thats what i thought, as IAX is spozed to be a nat friendly protocol |
| 12:28.32 | h3x | you basically had iax inbound disabled on your telehouse box |
| 12:28.36 | IP_Fox | i just did it for good measure |
| 12:28.36 | RoyK | there should have been a nat=auto |
| 12:28.45 | h3x | there really should be a nat=sucks |
| 12:28.55 | IP_Fox | yeah! |
| 12:29.02 | h3x | but i think its hard coded into asterisk |
| 12:29.09 | IP_Fox | :D |
| 12:29.15 | RoyK | as in "if ip is rfc1918, set nat=yes" |
| 12:29.40 | h3x | the thing is |
| 12:29.50 | h3x | it shouldnt be necessary to even tell it |
| 12:30.04 | IP_Fox | my thoughts exactly... (which i guess was y i never put it in!)( |
| 12:30.10 | h3x | it shoudl assume nat if the ip address specified in sip for the rtp port... hmmm |
| 12:30.11 | RoyK | you need to have some sort of nat handling with sip |
| 12:30.16 | h3x | dosent match the socket |
| 12:30.17 | h3x | well |
| 12:30.20 | h3x | i guess you cant do that |
| 12:30.23 | h3x | if the rtp is forwarded |
| 12:30.37 | h3x | err. |
| 12:30.40 | h3x | that dosent matter |
| 12:30.41 | IP_Fox | i guess assumption is the mother of all f**** ups! |
| 12:30.43 | RoyK | the sip header will contain the local NATed address |
| 12:30.46 | h3x | why dosent it just resolve it |
| 12:30.57 | RoyK | so trying to send something to 10.0.0.4 over the internet works badlyu..... |
| 12:31.06 | RoyK | badly, even |
| 12:31.08 | h3x | yeah but if its local |
| 12:31.13 | h3x | you cant go by the ip addressing shit |
| 12:31.18 | h3x | because if its on your LAN you dont want it to assume nat |
| 12:31.27 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 12:31.33 | h3x | thats the mistake they made in asterisk originally |
| 12:31.47 | RoyK | h3x: sure, but then allowing the "nat=detect" would make sense |
| 12:31.50 | RoyK | nat=rfc1918 |
| 12:31.52 | RoyK | perhaps |
| 12:32.14 | DrukenHME | why don't we all just agree NAT is a pain in the ass and shouldn't be used? :) |
| 12:32.20 | h3x | nah all ya need to do is compare the headers sent in SIP against the ip address it sees of the other endpoint on the berkeley sockets side |
| 12:32.33 | IP_Fox | sure, but we'd be buggered for IP's if we didnt use it! |
| 12:32.36 | h3x | IPV6 !#%#!%#! |
| 12:32.54 | h3x | I remember back in the good old days |
| 12:33.07 | h3x | I had a dedicated 33.6k modem with a whole Class C routed to it |
| 12:33.08 | DrukenHME | i don't think the shortage on ip's is as bad as everything thinks... |
| 12:33.14 | RoyK | DrukenHME: something like 95% of our customers are behind NAT |
| 12:33.14 | DrukenHME | i mean.... i managed to get 32... |
| 12:33.24 | RoyK | can't do anything about that |
| 12:33.33 | h3x | Well the thing they fucked up is assigning 2/5ths of the IPs to useless Class D and E |
| 12:33.35 | IP_Fox | ditto |
| 12:33.36 | DrukenHME | RoyK: it's still a pain in the ass :) |
| 12:33.47 | h3x | D is multicast and like 1 address is used |
| 12:33.51 | h3x | and E is experimental |
| 12:34.12 | RoyK | DrukenHME: still you gotta live with it |
| 12:34.20 | h3x | i got wiltel to give me two class C's with my DS3 |
| 12:34.22 | h3x | which is suprising |
| 12:34.54 | h3x | that remidns me i need to set up a phantom box that has all those ip aliases configed up so it looks like im using all the ip addresses |
| 12:35.08 | h3x | before the whatchamacallit audits it |
| 12:35.31 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 12:35.32 | DrukenHME | why not just use them ? |
| 12:35.43 | DrukenHME | no sence in hording ip's if you don't need them |
| 12:35.43 | h3x | heh |
| 12:35.54 | h3x | i will |
| 12:36.09 | h3x | for colo customers and more voip gateways eventually |
| 12:36.13 | DrukenHME | that's why there's a shortage... |
| 12:36.34 | DrukenHME | i may have 32 ip's... but i also use 32 ip's... |
| 12:37.01 | jontow | I have a single IP.. and it isn't static.. all of my IP space is at work.. and thats 4 IPs :P |
| 12:37.20 | h3x | it wont take me too long to run out of IPs once im in full production |
| 12:41.25 | *** join/#asterisk [shodan] (~shodan@216.113.99.186) |
| 13:06.34 | *** join/#asterisk tessier (~treed@210.245.38.7) |
| 13:14.33 | *** join/#asterisk smurfix (~smurf@smurfix.developer.debian) |
| 13:20.41 | jontow | crap |
| 13:20.51 | jontow | "Action: Queues" doesn't send a Response: line |
| 13:20.56 | jontow | no wonder i've been having problems with this proxy |
| 13:23.38 | *** join/#asterisk gpearson (~Graham@c-67-177-182-16.hsd1.in.comcast.net) |
| 13:23.53 | Makenshi | ipv6++ :> |
| 13:25.15 | *** join/#asterisk gpearson (~Graham@c-67-177-182-16.hsd1.in.comcast.net) |
| 13:38.40 | *** join/#asterisk mw` (~michael@omega.gc-schwartz.de) |
| 13:38.48 | L|NUX | can some one tell me from where i can get video codecs for asterisk ? |
| 13:40.23 | jontow | implement support for the ogg/vorbis libraries |
| 13:40.28 | jontow | you will be a hero to me |
| 13:41.44 | cjk | hi, whats the equivalent options for iax for canreinvite=yes ? |
| 13:43.16 | jontow | something akin to 'qualify=..' ? |
| 13:44.11 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 13:51.26 | cjk | hi, iax-phone1 is talking to iax-phone2, both using the same codec (gsm) but i see that the traffic passes through my * anyway to change this? they should talke directly to each other |
| 13:57.58 | roamer323 | cjk - ain't gonna happen - media is in the same pipe as control - iax2 is basically a trunking protocol - you can transfer the call, but you'll lose signal/control - i.e. no CDR/billing possible |
| 14:00.50 | *** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au) |
| 14:03.48 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:06.59 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-035-067.arcor-ip.net) |
| 14:07.22 | roamer323 | zeeek - r u in luv with yr polycom yet? |
| 14:07.54 | DrukenHME | one must have small penis to make love with telephone... |
| 14:08.56 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 14:09.43 | Zeeek | O don't have a polycom |
| 14:19.49 | marlowe | <DrukenHME> one must have small penis to make love with telephone... |
| 14:21.11 | *** join/#asterisk lbarth (user@pD9EA607A.dip.t-dialin.net) |
| 14:22.14 | *** join/#asterisk wiseguy_ (~chivilis@vadyba.vtu.lt) |
| 14:22.24 | wiseguy_ | helow |
| 14:22.54 | wiseguy_ | anyone with cisco peer? |
| 14:24.32 | DrukenHME | marlowe: well... wouldn't they ? |
| 14:24.38 | wiseguy_ | help me |
| 14:34.47 | Zeeek | what are you, some kind of wiseguy? |
| 14:35.32 | smurfix | Anybody know of some reason why * doesn't play my dial tone any more? (Other tones do play, zaptel-connected phone.) |
| 14:35.52 | Zeeek | TDM400 ? |
| 14:36.20 | smurfix | Zeeek: yep |
| 14:36.21 | Zeeek | try stopping asterisk, unloading and reloading wcfxs |
| 14:36.45 | Zeeek | I've had the dialtone drop once or twice, fortunately not often |
| 14:36.48 | smurfix | Zeeek: didn't work, even after reboot |
| 14:36.54 | Zeeek | eewww |
| 14:37.09 | Zeeek | drums stop. no good. |
| 14:37.50 | wiseguy_ | anyone using grandstream ip phones? |
| 14:37.57 | smurfix | what's so special about the dialtone anyway ..? |
| 14:37.59 | Zeeek | yes, millions |
| 14:38.03 | *** join/#asterisk gonzo- (~gonzo@icc-nat.univ.kiev.ua) |
| 14:38.28 | cjk | roamer323, so iax is in that point a looser compared to sip. imagine a firm with 2 offices in hongong an the head office in eu. hongkong1 calls hongkong2 and the traffic is travelling throuhg europe. thats shit |
| 14:39.09 | Zeeek | why is it travelling through europe? |
| 14:39.24 | cjk | Zeeek, there is the main * |
| 14:39.32 | cjk | everyone is registered there |
| 14:39.49 | cjk | Zeeek, with sip this wont be a problem |
| 14:39.57 | Zeeek | it can go direct after the connection is established |
| 14:40.09 | cjk | Zeeek, ok tell me how |
| 14:40.26 | Zeeek | I don't know how, but just last friday I saw bridging direct on one of mine |
| 14:40.40 | cjk | Zeeek, that means it does not do transcoding |
| 14:40.56 | cjk | but the traffic is passing through * |
| 14:41.11 | Zeeek | I'm not sure you are right about this. There is an equivalent of "canreinvite" in iax tho I don't recall what it is |
| 14:41.23 | cjk | it is called notransfer |
| 14:41.27 | cjk | but i will loose the cdr record |
| 14:41.30 | cjk | imagine this |
| 14:41.39 | Zeeek | ah, you are talking about two different things now |
| 14:42.09 | Zeeek | if it'sz two offices talking directn they may not need the cdr. If it's billing you want that's a different point |
| 14:42.12 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 14:42.24 | cjk | well one needs both |
| 14:42.35 | Zeeek | tough titty perhaps |
| 14:43.45 | cjk | well impossible |
| 14:43.51 | cjk | thats really a shitty thing about iax |
| 14:44.02 | smurfix | Zeeek: it even says "WARNING[1250]: Unable to play dialtone on channel 1" in the log. Doesn't say why though :-/ |
| 14:44.10 | Zeeek | I am not competent to answer this |
| 14:44.30 | smurfix | time to dig through the sources ... |
| 14:44.33 | Zeeek | smurfix first I've heard of that? What did you do lately? Recompile |
| 14:44.34 | Zeeek | ? |
| 14:44.54 | Zeeek | cjk I am not competent to answer this |
| 14:45.40 | *** join/#asterisk cybast (~cybast@64-235-222-179.access.ripnet.com) |
| 14:46.04 | smurfix | Zeeek: No, switch indications to German. That worked. Then I rebooted (for entirely different reasons). |
| 14:46.26 | smurfix | Zeeek: After that, no more tone. |
| 14:46.31 | cybast | I am a nebie to asterisk and am wonderin gif someone could answer a question for me |
| 14:46.34 | Zeeek | smurfix so now it works, or that's how you broke it? |
| 14:46.39 | wiseguy_ | help me someone with grandstream ip phone |
| 14:46.41 | Zeeek | ask cybast |
| 14:46.43 | smurfix | That's how I broke it. |
| 14:46.48 | wiseguy_ | i can't get it to register |
| 14:46.59 | Zeeek | smurfix aha, then your indications file is bad somehow |
| 14:47.01 | wiseguy_ | Apr 3 17:46:53 NOTICE[32178]: chan_sip.c:7691 handle_request: Registration from '<sip:oper@10.10.10.2>' failed for '10.10.10.102' |
| 14:47.23 | Zeeek | wiseguy password or username problem |
| 14:47.43 | wiseguy_ | Zeeek i have tried with no password |
| 14:47.47 | wiseguy_ | and no username |
| 14:47.50 | wiseguy_ | the same shit |
| 14:48.15 | cybast | I have a question regarding the extensions.conf file re: incoming calls on a Digium card |
| 14:48.23 | Zeeek | put your sip.conf entry for the phone in htp://pastebin.ca |
| 14:48.35 | Zeeek | cybast ask the question, don'tdescribe it |
| 14:48.39 | cybast | ok |
| 14:48.58 | *** join/#asterisk VirTERM (~VirTERM@204.225.113.90) |
| 14:49.00 | NewSole | anybody hear have a clear voice for recording prompts |
| 14:49.05 | VirTERM | morning |
| 14:49.29 | wiseguy_ | http://pastebin.ca/8719 |
| 14:49.32 | wiseguy_ | Zeeek |
| 14:49.46 | tessier | NewSole: I can lend you the talents of my Marge Simpson impersonation |
| 14:49.56 | smurfix | NewSole: you definnition of "clear" or mine ;-) |
| 14:50.00 | smurfix | s/nn/n |
| 14:50.13 | cybast | I have a TDM31B card from digium with 3 extensions in my house. I can make local calls to extensions within my house and have set up the extensions.conf to handle outgoing calls on the trunk however I can't seem to figure out what happens when an incoming call comes in on the trunk. How do I direct the call to one or all of the extensions? |
| 14:50.16 | Zeeek | wiseguy_ remove host and fromuser |
| 14:50.30 | Zeeek | and username |
| 14:50.58 | Zeeek | and all codecs except ulaw |
| 14:51.11 | Zeeek | and make sure the GS is set to ulaw |
| 14:51.22 | cybast | rebiit |
| 14:51.55 | Zeeek | what trunk? You mean you have analog phone lines? |
| 14:52.02 | cybast | yes |
| 14:52.06 | Zeeek | what happens when the call comes in? |
| 14:52.08 | cybast | al lanalog at present |
| 14:52.11 | Zeeek | on the console? |
| 14:52.17 | wiseguy_ | Zeeek what sip User ID put in the GS config? |
| 14:52.19 | cybast | thats what I'm not sure of |
| 14:52.34 | Zeeek | cybast are you using @home or AMP or something? |
| 14:52.50 | cybast | I've done alot of reading and can't seem to find info on how to handle the calls coming in on the analog trunk |
| 14:53.23 | Zeeek | cybast if you've read any of these you'd know: |
| 14:53.24 | Zeeek | Starter tutorial: |
| 14:53.24 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 14:53.24 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 14:53.24 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 14:53.24 | Zeeek | THE reference of the moment: |
| 14:53.26 | Zeeek | http://www.asteriskdocs.org |
| 14:53.54 | cybast | I was using asterisk@home v0.7 but was on here the other noght talking to VirTERM and couldn't seem to get anything working so I switched to fedora core 1 with the latest cvs tree of asterisk |
| 14:54.13 | cybast | I've done a ton of readings but can't seem to find info on incoming trunk configs |
| 14:54.18 | Zeeek | the second and third links above both have your answers |
| 14:54.28 | cybast | ok thanks I'll ook agian |
| 14:54.39 | cybast | thanks for the help |
| 14:54.47 | Zeeek | np, good luck |
| 14:55.17 | cybast | thanks |
| 14:56.33 | Darwin[laptop] | wow I lost an hour |
| 14:56.48 | Zeeek | you'll get it back in a few months |
| 14:57.51 | Darwin[laptop] | ~docs |
| 14:57.52 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:58.23 | VirTERM | realtime question: do I have to keep include statements in extensions.conf? How would you do it in SQL? |
| 14:59.10 | Dovid | hi all |
| 14:59.19 | Dovid | i just installed asterisk |
| 14:59.57 | Dovid | anyone know where i can get simple commands so i cant set up a mini test pbx. i want to e able to connetc to phones to the pbx and they whould be able to call each other |
| 15:00.03 | smurfix | Hmm. "ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device". That sounds about the appropriate error. Now if it wouldn't get written to /dev/null people could actually see it. |
| 15:00.09 | VirTERM | wiki... |
| 15:00.24 | Dovid | link ? |
| 15:00.37 | VirTERM | <PROTECTED> |
| 15:00.39 | Darwin[laptop] | ~docs |
| 15:00.40 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:00.45 | Darwin[laptop] | start there |
| 15:03.43 | VirTERM | cybast: still having problem with incoming calls?!? |
| 15:04.40 | cybast | just reading now |
| 15:04.52 | VirTERM | ok |
| 15:04.53 | cybast | Thanks for all the help the other noght |
| 15:04.55 | cybast | night |
| 15:05.13 | cybast | I decided to load Fedora and the config files from scratch rather than using asterisk@home |
| 15:05.17 | VirTERM | well, it seems that you still have some problems... |
| 15:05.34 | Zeeek | * is a sign give to restaurants in the Michelin guide, the more * the better the restaurant |
| 15:05.34 | VirTERM | are you using CVS or 1.0.7? |
| 15:05.36 | cybast | I now have calls between internal extensions working and outgoing calls on the trunk working |
| 15:06.13 | cybast | I donwload the latest cvs tree 1.0.7 i think |
| 15:06.47 | VirTERM | 1.0.7 is the current released version, but it doen's really matter |
| 15:08.04 | cybast | I am just trying to get the incoming trunk to ring all the extensions |
| 15:08.15 | cybast | then i want to set up time restrictions . .etc |
| 15:08.23 | Zeeek | keep reading! |
| 15:08.24 | cybast | starting off simple with the extensions.conf file |
| 15:08.48 | cybast | yes . . back to reading . .the page you referred to seems to have the answer |
| 15:08.56 | Zeeek | they both do |
| 15:10.01 | Darwin[laptop] | asterisk@home is a pain |
| 15:10.02 | Zeeek | the line in zapata.conf "context=" under the channel for the FXO will determine where the incoming call goes. Under that context in extensions.conf, you will then use the 's' extension to Dial() the phones you want to ring. It('s all there |
| 15:10.08 | VirTERM | tip: need to define context in zapata which has coresponding entry in extensions.conf... |
| 15:10.20 | Darwin[laptop] | I have tried it on 4 x86 boxes and it failed to install on them all |
| 15:10.40 | roamer323 | cjk: iax2 and sip are designed for very different purposes - but people just insist on comparing them at par - the existence of IAXy and iax phones do not help the cause |
| 15:11.16 | Darwin[laptop] | iax2 is a better protocal |
| 15:11.19 | Darwin[laptop] | sip has to many nat issues where iax2 does not |
| 15:11.24 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 15:11.35 | Darwin[laptop] | and its great for trunking *boxes together |
| 15:11.39 | Zeeek | actually, iax does have some if you try to use multiple clients behind one NAT |
| 15:19.50 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 15:20.42 | *** join/#asterisk bah (048830696@ACA569D5.ipt.aol.com) |
| 15:23.54 | cybast | another quick question guys . . . in the article that zeeek referred me to it says that the digium card should have it's own IRQ all to itself. I have moved the card in to every pci slot in my machine and it seems that no matter where I place it, it alway sends up sharing an interrupt with something else. |
| 15:24.58 | VirTERM | check your BIOS if you can reserve IRQs for certain PCI slots |
| 15:25.25 | cybast | I have a section that has IRQ reservations but it doesn't put it against a slot |
| 15:26.08 | cybast | I have a dell optiplex GX110 |
| 15:26.33 | Zeeek | cybast you'll have to remove any unneeded stuff if possible |
| 15:26.49 | Zeeek | read about the IRQ issue here: http://asteriskdocs.org |
| 15:27.39 | cybast | most of the stuff is integrated into the mainboard and most of it can't be turned off (other than the nic card which of course I don't want off) |
| 15:27.54 | cybast | I'll read about the IRQ issue |
| 15:27.57 | cybast | thanks |
| 15:28.13 | cjk | Darwin[laptop], oncee the nat issues are solved, sip is better. and in the future every nat will have sip-pass-through options |
| 15:28.18 | Zeeek | gotta be a way! what about USB? You need that? |
| 15:28.37 | Zeeek | cybast^^^ |
| 15:29.06 | cybast | I can;t turn the usb off |
| 15:29.12 | VirTERM | but shared IRQ issue should not stop you from getting incoming calls to work.. |
| 15:29.20 | cybast | there is nowhere in the bios to do that |
| 15:29.21 | Zeeek | I'd check on all this in the Dell Usenet group or webforum |
| 15:29.35 | Zeeek | shared IRQ is almost guaranteed to fuck things up though |
| 15:29.47 | cybast | no . . i should still be able to get calls workign I just noticed that when reading the article that zeeek told me about |
| 15:29.54 | VirTERM | one thing at a time... |
| 15:30.02 | Zeeek | yes, but people often complain about clickings and scratchings |
| 15:30.10 | cybast | I'll find out more info on the irq before tackling the incoming call thing |
| 15:33.02 | Zeeek | cybast: http://lists.suse.com/archive/suse-linux-e/2002-Jun/3352.html |
| 15:33.19 | Zeeek | The Dell Optiplex |
| 15:33.19 | Zeeek | GX110 slimline PCs share IRQs between the USB controller and the network |
| 15:33.19 | Zeeek | card. |
| 15:33.20 | cybast | thanks zeeek . . I'll go take a look now |
| 15:33.44 | cybast | I don't have a slimline I have a tower |
| 15:33.45 | Zeeek | this states you can turn off USB which shares an IRQ with the NIC! |
| 15:33.53 | Zeeek | smae idea I'm sure |
| 15:34.15 | cybast | I was on the dell site and it looks like there might be bios updates |
| 15:34.20 | Zeeek | this does not look like an excellent asterisk box |
| 15:34.26 | cybast | really |
| 15:34.50 | Zeeek | try also googling the mailing list for that model - maybe someone has already been there |
| 15:35.06 | cybast | From the articles I read It seemed dell was a good way to go for supported hardware so I bought this on ebay specifically for an asterisk box |
| 15:35.08 | jontow | those Slimlines suck in general :/ |
| 15:35.17 | jontow | they have full duplex sound issues too |
| 15:35.21 | cybast | it's not a slimline . . . it's a tower |
| 15:35.25 | jontow | i know. |
| 15:35.41 | jontow | we have 3 of them here that were used as agent PCs.. they were the newest in the room, and the first to be decommissioned |
| 15:35.44 | Zeeek | the optiplex line is for offices ? |
| 15:36.14 | cybast | I think so |
| 15:37.23 | Zeeek | cybast type this into google search window: optiplex site:lists.digium.com |
| 15:38.22 | *** join/#asterisk Silik0n (~krice@rso.suspicious.org) |
| 15:39.31 | cybast | ok |
| 15:40.24 | Zeeek | well, I think you have enough homework for today :) |
| 15:40.41 | cybast | indeed I do. |
| 15:40.45 | cybast | more reading |
| 15:41.23 | Zeeek | someone said the Optiplex GX1 "makes a great linux and asterisk box" and goes for $35 on e-bay |
| 15:41.24 | Silik0n | anyone got * to compile on obsd lately |
| 15:41.38 | Silik0n | GX1s reant bad boxen... |
| 15:41.48 | Silik0n | i wouldnt wanna do any real number crunching on them tho |
| 15:42.06 | Zeeek | well, for $35 what d'ya want ? |
| 15:42.34 | Silik0n | webserver, firewall, nfs server |
| 15:43.56 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-104-43-8.buff.east.verizon.net) |
| 15:44.28 | SuPrSluG | hi all |
| 15:45.37 | *** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net) |
| 15:47.07 | VirTERM | http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&category=56101&item=5763805588&rd=1&ssPageName=WD4V |
| 15:47.12 | VirTERM | nice * machine |
| 15:48.29 | dieck | 1 HE? so max. 1 additional (e.g. pci) card? |
| 15:48.44 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 15:48.46 | VirTERM | quad pri |
| 15:49.00 | VirTERM | you won't get much more from a single box anyway |
| 15:49.37 | dieck | hm, ok, quad pri is like more than one card :) |
| 15:50.15 | JonR800 | lol.. I run Asterisk on a GX1 as well |
| 15:51.36 | VirTERM | quad pri is a single PCI card which allows you to connect up to 4 pri lines |
| 15:54.33 | Silik0n | you might run * on a GX1 but does it do any transcoding? |
| 15:55.05 | Silik0n | and the problem with more then 1 wct4xxp in a box sint cpu time, its interupt service time |
| 15:55.55 | JonR800 | sure it does some transcoding, it's just a small box, doesn't handle massive call volume. If it did, it wouldn't be a GX1 :) |
| 15:59.17 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 15:59.38 | file[laptop] | yay oej |
| 15:59.47 | oej | yay |
| 16:00.18 | Silik0n | oej: who's incharge of the obsd port? |
| 16:00.34 | file[laptop] | oej: how are you? |
| 16:00.41 | Silik0n | what up file |
| 16:00.52 | file[laptop] | not me considering I haven't gotten out of bed yet |
| 16:00.56 | file[laptop] | I probably should... |
| 16:01.01 | Silik0n | hah |
| 16:01.28 | Silik0n | ENDIAN detection for OBSD is fuxored |
| 16:01.37 | zoa | coming soon |
| 16:01.38 | zoa | :) |
| 16:02.12 | Silik0n | and app_queue is b0rked too |
| 16:02.26 | file[laptop] | that's old news |
| 16:03.05 | file[laptop] | okay gotta force myself to get out of bed |
| 16:03.09 | file[laptop] | and make food, and setup this new box |
| 16:03.15 | file[laptop] | that took 9 e-mails to get, and 2 phone calls... yow'sa |
| 16:03.17 | Silik0n | it error[1]'s on obsd completely |
| 16:03.51 | Silik0n | note to self just fix the shit yourwelf and put a patch on mantis |
| 16:05.28 | blitzrage | file[laptop]: I have to do the exact same thing! |
| 16:05.45 | tzanger | so much for a stealthy entrance |
| 16:05.54 | blitzrage | tzanger: *gasp* |
| 16:06.00 | blitzrage | tzanger: I've killed men for less |
| 16:06.14 | blitzrage | tzanger: luckily you're no man (OH!) |
| 16:06.38 | tzanger | heh |
| 16:06.49 | blitzrage | so whats shakin? |
| 16:07.12 | tzanger | nada gotta go to crappy tire in this shit to get a replacement cartridge and a couple o-rings for my kitchen faucet |
| 16:07.19 | tzanger | fixed the vacuum cleaner yesterday |
| 16:07.24 | smurfix | I understand you can talk to ISDN with zaphfc, mISDN, or CAPI. So what's actually better? |
| 16:07.25 | tzanger | now I just need to fix the transmission |
| 16:07.28 | tzanger | not a man... phsaw |
| 16:08.00 | smurfix | (no NT mode necessary) |
| 16:08.43 | tzanger | http://64.236.34.67:80/stream/1017 is currently what's playing |
| 16:08.55 | tzanger | kind of bland but they had some good stuff on a minute ago |
| 16:09.31 | blitzrage | fix the transmission? that sounds slightly more complicated than the cartridge and o-rings |
| 16:10.01 | tzanger | blitzrage: it's not bad, there's a solenoid on this particular one that is always going, it's a known weak part |
| 16:10.08 | tzanger | it's got it's own access cover and everything |
| 16:10.18 | blitzrage | tzanger: wow! get a stronger solenoid |
| 16:10.29 | blitzrage | hrmmmm.... |
| 16:10.31 | tzanger | however I have to take it in, as there's a boost valve that also needs to be replaced and that, unfortunately, is buried in the transmission |
| 16:10.37 | Silik0n | Beginning asterisk shutdown.... |
| 16:10.37 | Silik0n | asterisk in free(): error: chunk is already free |
| 16:10.37 | Silik0n | Abort trap (core dumped) |
| 16:10.47 | blitzrage | oh I know why |
| 16:10.49 | tzanger | blitzrage: that ought to be fun |
| 16:10.50 | Silik0n | i love it when it does that |
| 16:10.51 | blitzrage | double NAT! |
| 16:10.59 | Silik0n | double nat ++++ |
| 16:11.12 | blitzrage | I'm kind of doing a wierd topology... |
| 16:11.16 | *** join/#asterisk iq (~iq@207-224-100-90.omah.qwest.net) |
| 16:11.37 | blitzrage | the phone goes out one router and loops back in on another IP at a different router in the same physical location |
| 16:12.30 | blitzrage | aha! and the tftp server IP is wrong in the phone |
| 16:12.33 | blitzrage | that'll do it |
| 16:12.41 | *** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net) |
| 16:12.55 | blitzrage | file[laptop]: lol |
| 16:12.55 | tzanger | file[laptop]: that's a quick way to lose it |
| 16:13.10 | file[laptop] | nah |
| 16:13.12 | file[laptop] | it's pretty stable |
| 16:13.59 | JerJer[interop] | anyone using -stable: there are reports that something has changed with the relaxdtmf option from 1.0.5 to 1.0.7. Anyone have any ideas? |
| 16:14.25 | blitzrage | JerJer[interop]: have you checked CVS? :D |
| 16:14.36 | tzanger | blitzrage: hahahahahah |
| 16:14.42 | tzanger | how long have you waited to zing him with that |
| 16:14.51 | blitzrage | tzanger: about 2 years |
| 16:15.01 | facek_ | hi |
| 16:15.23 | facek_ | what I need to connect asterisk with my cellular phone or only simcard and use that as channel for making calls. |
| 16:15.31 | tzanger | you two need to get a room |
| 16:15.41 | blitzrage | tzanger: this is a room |
| 16:15.43 | file[laptop] | care to join us? |
| 16:15.50 | facek_ | tzanger room? |
| 16:16.00 | tzanger | hahaha |
| 16:16.02 | tzanger | no and no |
| 16:16.17 | file[laptop] | c'mon |
| 16:16.20 | facek_ | yyy? |
| 16:20.57 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 16:22.13 | blitzrage | anyone know why a 7960 running 7.3 firmware doesn't accept start_media_port: "16000" in the <mac>.cnf file? still has the default configured on the phone |
| 16:25.48 | maik | I think it has to be SIP<mac>.cnf |
| 16:26.05 | cybast | zeeek and VirTERM - IRQ Problem solved. Had to update bios from A05 to A09 which let me turn off the onboard usb controller . . moved digium card to pci slot4 which shares IRQ with USB and voila . . . now onto incoming trunk calls |
| 16:26.07 | blitzrage | maik: right. I have that, the file is being read, thats not the problem :) |
| 16:26.15 | blitzrage | hehe |
| 16:26.49 | file[laptop] | it just hates you blitz |
| 16:27.02 | blitzrage | tell me about it |
| 16:27.08 | file[laptop] | just like 'da server |
| 16:27.14 | blitzrage | I shall |
| 16:27.41 | JerJer[interop] | blitzrage: har ar |
| 16:27.43 | JerJer[interop] | har |
| 16:27.49 | JerJer[interop] | <-- notice nic |
| 16:27.51 | JerJer[interop] | nick |
| 16:27.55 | blitzrage | JerJer[interop]: glad I could help! :) |
| 16:28.13 | blitzrage | I'm not really sure what it means... |
| 16:29.01 | VirTERM | cybast: good progress...keep going at this rate and we gonna have you up and running in 1 hr |
| 16:29.28 | cybast | cheers! |
| 16:32.53 | cybast | VirTerm -> In my zapata.conf file I have the am econtext defined for the 3 fxs and the fxo . .is this ok? |
| 16:33.02 | cybast | same context |
| 16:33.37 | JerJer[interop] | blitzrage: i am at the network world interop testing right now |
| 16:34.08 | JerJer[interop] | working with asterisk and sip |
| 16:38.35 | VirTERM | you should really have to seperate contexts; on for FXS channels (sort of trusted) and one for your FXO |
| 16:38.50 | VirTERM | but for testing purposes one context will do |
| 16:39.37 | VirTERM | now, you need to handle an incoming call from FXO in your extensions.conf |
| 16:39.49 | VirTERM | for testing you can just ring one of your FXS ports |
| 16:40.07 | VirTERM | exten=> s,1,Answer |
| 16:40.35 | VirTERM | exten => s,2,Dial(Zap/YOUR CHANNEL HERE|20) |
| 16:40.55 | cybast | I will create a new context then for the trunk. Do I put the context line above the signalling and channel lines |
| 16:40.58 | VirTERM | then "extensions reload" |
| 16:41.14 | cybast | cool thanks |
| 16:41.33 | cybast | I had better do it the right way with 2 contexts |
| 16:41.53 | VirTERM | anywhere within the configuration of specific trunk |
| 16:42.36 | VirTERM | just look at the example in /usr/src/asterisk/configs |
| 16:43.33 | *** join/#asterisk moy (~moy@201.128.210.194) |
| 16:43.38 | cybast | in the zapata.conf file though right?? So I specify one centext for the 3 fxs lines and an connxt for fxo and conxt line comes before he channel def |
| 16:43.49 | VirTERM | yes |
| 16:43.58 | cybast | ok . . . away I |
| 16:44.02 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 16:44.02 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 16:44.04 | cybast | away i go |
| 16:44.10 | *** mode/#asterisk [-r] by bkw_ |
| 16:44.17 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released |
| 16:44.23 | cybast | thanks |
| 16:44.27 | blitzrage | JerJer[interop]: I'm trying to debug why my Asterisk keeps sending OPTIONS Retransmits to a phone... |
| 16:46.43 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 16:47.11 | mw` | whats the difference between Congestion and Busy? (sorry im from germany) |
| 16:47.25 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 16:47.38 | file[laptop] | blitzrage: is it receiving a response? |
| 16:48.34 | blitzrage | file[laptop]: nope, the options are obviously Asterisk sending the "ping" to the phone, but the phone must not be seeing them to reply. I can verify with a packet sniff, but I need to get access to the other box, which my roommate controls :) |
| 16:48.54 | VirTERM | Congestion = no available channels |
| 16:49.50 | mw` | VirTERM: ok thanks :) |
| 16:49.52 | JerJer[interop] | yay - i get to install and configure SER today |
| 16:50.08 | blitzrage | JerJer[interop]: YES! |
| 16:50.59 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
| 16:51.57 | JerJer[interop] | it will be interesting, that's for sure |
| 16:52.09 | blitzrage | yes.... interesting |
| 16:54.27 | bkw_ | JerJer[interop], might as well tell everyone people fell for the asterisk 2.0 joke |
| 16:54.37 | JerJer[interop] | strangely enough so far i like SER |
| 16:55.00 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 16:55.07 | JerJer[interop] | bkw_: yes i had 4 seperate people come up to me asking questons about asterisk 2.0 |
| 16:55.34 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 16:55.59 | JerJer[interop] | but i think it was really one guy that very quickly read that mailing list post, then the others had heard about stuff from listening to the dev conferences, so they assumed it was true |
| 16:56.19 | bkw_ | hahaha |
| 16:56.20 | bkw_ | could be |
| 16:56.21 | file[laptop] | bkw_: I've come aross something in logger.c that makes no sense to me, I think someone mistyped something... |
| 16:56.28 | bkw_ | file shoot |
| 16:56.32 | file[laptop] | line 609 |
| 16:56.42 | file[laptop] | look at the if statement below it for src... |
| 16:56.57 | file[laptop] | shouldn't that be str? |
| 16:57.28 | JerJer[interop] | rut ruh |
| 16:57.31 | bkw_ | if !str |
| 16:57.35 | bkw_ | is what it should be |
| 16:57.37 | file[laptop] | yeah |
| 16:57.39 | file[laptop] | I thought so |
| 16:57.47 | bkw_ | typo |
| 16:57.58 | bkw_ | the new strip color stuff hahahahah |
| 16:58.00 | zoa | AHA! |
| 16:58.15 | file[laptop] | sounds like an... oops! |
| 16:58.26 | bkw_ | ok app_websms supports 6 providers I should post it so people can fix and expand it |
| 16:58.37 | file[laptop] | but not Telus :( |
| 16:58.53 | JerJer[interop] | http://pastebin.ca/8726 <--- seg fault in libpri |
| 16:58.55 | bkw_ | you can figure that out and add it |
| 16:59.02 | facek_ | where Can i find PGSQL adds? |
| 16:59.18 | bkw_ | facek_, their isn't any |
| 16:59.21 | bkw_ | use odbc |
| 16:59.50 | JerJer[interop] | bkw_: att/cingular ? |
| 16:59.59 | bkw_ | yes |
| 17:00.01 | facek_ | bkw_ I saw some time ago. i want to use in extensions.conf a simple select like exten => 1,1,PGSQL("SELECT .... |
| 17:00.02 | JerJer[interop] | kick ass |
| 17:00.09 | JerJer[interop] | facek_: no you don't |
| 17:00.11 | bkw_ | its just a web api to smack the web form |
| 17:00.18 | bkw_ | for the cli |
| 17:00.20 | bkw_ | and from the dialplan |
| 17:00.28 | JerJer[interop] | facek_: wriite an app |
| 17:00.28 | facek_ | JerJer[interop] so how, can i get data in extensions from database? |
| 17:00.36 | facek_ | JerJer[interop] what app? |
| 17:00.44 | JerJer[interop] | write one |
| 17:00.58 | JerJer[interop] | exten => 1,1,CollectData() |
| 17:01.00 | JerJer[interop] | whatever |
| 17:01.02 | file[laptop] | I'm sorry dear facek, but everything isn't going to be handed to you on a silver platter |
| 17:01.04 | facek_ | which one? that app exists |
| 17:01.09 | JerJer[interop] | use the power of asterisk, don't sidestep it |
| 17:01.18 | facek_ | i want to get from database the name of calling client |
| 17:01.29 | facek_ | and set that as CALLERIDNAME |
| 17:01.31 | JerJer[interop] | then do that |
| 17:02.05 | facek_ | how? |
| 17:02.43 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 17:04.00 | facek_ | hm |
| 17:04.53 | JerJer[interop] | w r i t e an ap |
| 17:05.00 | ManxPower | facek_: You have to learn to walk before you run. See "show application DBGet" |
| 17:05.11 | ManxPower | Also "show application AGI" |
| 17:05.21 | ManxPower | And hell, you might as well run "show applications" too. |
| 17:05.29 | ManxPower | ~docs |
| 17:05.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:05.33 | ManxPower | ~mailinglist |
| 17:05.34 | 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/ |
| 17:05.34 | JerJer[interop] | ManxPower: have you played with the relaxdtmf option on -stable? |
| 17:05.36 | facek_ | ManxPower about AGI i know.. but i don;t like it for that. |
| 17:05.41 | ManxPower | JerJer[interop]: Yes. |
| 17:05.58 | facek_ | soem time ago i see PGSQL in actin and it work, so what.. now it's doesn't work? |
| 17:06.32 | ManxPower | JerJer[interop]: For the most part relaxdtmf=yes causes subtle issues, like "3324" being seen as "324" |
| 17:06.45 | *** join/#asterisk iq[laptop] (~iq@207-224-100-90.omah.qwest.net) |
| 17:06.55 | cjk | hi, i now see the pro's are in. maybe you can tell me if its possible with iax to get the corrects cdr's and make the traffic not going through *, like sip canreivte=yes |
| 17:06.56 | *** join/#asterisk mrScamp|away (~sasha@hacker.pibhe.com) |
| 17:07.10 | file[laptop] | no. |
| 17:07.17 | ManxPower | cjk: see the "notransfer" option. |
| 17:07.32 | JerJer[interop] | ManxPower: we have reports that it has stopped functioning (at least as expected) between 1.0.5 and 1.0.7 |
| 17:07.49 | ManxPower | cjk: One of IAX2's dirty little secrets is that if the audio does not go thru Asterisk, the CDRs will be wrong. |
| 17:07.55 | cjk | ManxPower, i checke that out. put it into my iax file didnt change anything. set to yes or to no. so im a litte confused |
| 17:08.05 | ManxPower | JerJer[interop]: I don't set it because it causes ittus. |
| 17:08.08 | cjk | ManxPower, thats a really dirty one |
| 17:08.19 | JerJer[interop] | ittus ? |
| 17:08.26 | file[laptop] | it's more like, if the signalling doesn't go through asterisk... the CDRs will be wrong |
| 17:08.31 | ManxPower | JerJer[interop]: On first cup of coffee. "issues" |
| 17:08.51 | JerJer[interop] | iax simply does not seperate audio stream and signalling |
| 17:08.54 | ManxPower | file[laptop]: Yes, but since there is no difference between audio and signalling in IAX2... |
| 17:08.56 | facek_ | bkw_ but i don't want to look for CID name in asterisk database, i want to look in other postgresql database |
| 17:09.01 | blitzrage | ok, newb question! what packet trace program can I use to see the SIP packets (the contents) in the console? |
| 17:09.03 | file[laptop] | indeed yay |
| 17:09.11 | JerJer[interop] | once you deal with that ittus, everthing is fine |
| 17:09.15 | ManxPower | It's one of the VERY FEW bad things about keeping signaling and audio in the same stream. |
| 17:09.29 | cjk | ManxPower, a really bad one |
| 17:09.31 | ManxPower | blitzrage: "sip debug" |
| 17:09.36 | file[laptop] | need... partial... native transfers... |
| 17:09.40 | cjk | with cdr's are wrong do you mean wrong or missing |
| 17:09.49 | blitzrage | ManxPower: no, not the Asterisk CLI, Linux console. |
| 17:09.56 | blitzrage | ManxPower: I need to debug the other end. |
| 17:09.59 | ManxPower | cjk: the solution is not to allow IAX2 tranfers |
| 17:10.07 | ManxPower | blitzrage: tcpdump, ethereal |
| 17:10.13 | cjk | then i have the traffic |
| 17:10.16 | blitzrage | ManxPower: what is the option to see the packet contents? |
| 17:10.33 | JerJer[interop] | ManxPower: or the solution is to deal with the billing operation differently |
| 17:10.42 | MarkS_ | SOMEONE- Do recordings for my PBX?!?!! |
| 17:10.43 | ManxPower | blitzrage: tcpdump -X poprt 4569 |
| 17:10.43 | cjk | do you guys think iax will change in the far future and maybe support such an option |
| 17:10.50 | blitzrage | ManxPower: thanks, trying |
| 17:10.54 | file[laptop] | yay oej |
| 17:10.58 | MarkS_ | <PROTECTED> |
| 17:11.06 | ManxPower | MarkS_: Fuck off. |
| 17:11.09 | blitzrage | MarkS_: stop that |
| 17:11.16 | JerJer[interop] | where is an @ when we need one |
| 17:11.18 | file[laptop] | how rude! |
| 17:11.45 | MarkS_ | WTF |
| 17:11.47 | ManxPower | file[laptop]: I think bkw_ is getting all hot and sweaty with a SMS phone. |
| 17:11.47 | MarkS_ | .. |
| 17:11.52 | *** part/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 17:11.52 | file[laptop] | probably. |
| 17:11.56 | ManxPower | MarkS_: We don't take kindly to demands here. |
| 17:12.21 | ManxPower | Well THAT was easy. |
| 17:12.28 | file[laptop] | almost... too easy... |
| 17:12.33 | ManxPower | file[laptop]: Yeah. |
| 17:12.53 | file[laptop] | okay everyone, into the bomb shelter! |
| 17:12.59 | blitzrage | ManxPower: hrmmm... doesn't really show the whole packet.... only up to about the Via: |
| 17:13.10 | ManxPower | blitzrage: add a -s 2048 |
| 17:13.20 | *** join/#asterisk mrScamp|away (~sasha@hacker.pibhe.com) |
| 17:13.20 | blitzrage | ManxPower: you're the man :) |
| 17:13.29 | ManxPower | blitzrage: you know how you can thank me. 8-) |
| 17:13.57 | blitzrage | ManxPower: and to think, I was going to man tcpdump |
| 17:14.16 | ManxPower | blitzrage: All the info is on the man page, but the man page is...less then clear. |
| 17:14.28 | ManxPower | ...less THAN clear. |
| 17:14.37 | file[laptop] | silly ManxPower |
| 17:14.41 | cjk | sorry for asking again, but my question got maybe lost into marks's lines. do you guys think iax will change in the far future and maybe support such a "canreinvite" option. is there any small hope? |
| 17:14.45 | blitzrage | ManxPower: agreed. I'm looking at the options now... si there a way to NOT show the hex? :) |
| 17:14.56 | file[laptop] | cjk: maybe. |
| 17:15.05 | blitzrage | cjk: isn't it notransfer=yes ? |
| 17:15.08 | ManxPower | blitzrage: not that I know of, but I'm sure there is. |
| 17:15.19 | cjk | blitzrage, no really |
| 17:15.22 | file[laptop] | blitzrage: did I not teach you about IAX2 native transfers? :( |
| 17:15.29 | tzanger | wow |
| 17:15.33 | *** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net) |
| 17:15.33 | blitzrage | file[laptop]: yes... obviously I've forgotten.... |
| 17:15.33 | tzanger | there's an endorsement for Moen faucets |
| 17:15.34 | ManxPower | Fortunatly, I don't need to deal with CDRs yet (and not in the near future either) |
| 17:15.38 | tzanger | the cartridge and o-rings were totally free |
| 17:15.43 | blitzrage | tzanger: I used to work at Moen. |
| 17:15.47 | tzanger | in fact I spent an extra 10 minutes at Cdn Tire just trying to get a price check |
| 17:15.50 | blitzrage | tzanger: oh yah, I could have told you that |
| 17:16.00 | file[laptop] | blitzrage: with an IAX2 native transfer the entire call propogates off the box to go direct between the other two, so you never get accurate CDRs |
| 17:16.04 | tzanger | blitzrage: so why didn't you |
| 17:16.08 | blitzrage | tzanger: you don't have to pay for those, its part of the faucet for life |
| 17:16.13 | file[laptop] | blitzrage: he doesn't want that. |
| 17:16.16 | blitzrage | tzanger: I didn't know you were getting them for a Moen :) |
| 17:16.26 | patdk | heh, luckilly all my iax2 stuff is notransfers anyways |
| 17:16.36 | blitzrage | file[laptop]: see, the thing is that I thought notransfer=yes did that :) |
| 17:16.40 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 17:16.42 | tzanger | blitzrage: a likely excuse |
| 17:16.46 | cjk | ok guys. so i will see the cdr, source destination but i wont see the billsecs and duration |
| 17:16.52 | cjk | i can live with that |
| 17:16.54 | blitzrage | tzanger: look at the logs, you never told me what kind of faucet :) |
| 17:16.54 | blitzrage | lol |
| 17:16.55 | tzanger | reming me to kick you ass at the next torastricon |
| 17:17.01 | file[laptop] | blitzrage: it keeps the call on, but the entire call goes through... he just wants audio to go direct |
| 17:17.07 | blitzrage | tzanger: ok, but I might forget |
| 17:17.10 | tzanger | blitzrage: don't cloud this discussion with the facts! |
| 17:17.16 | blitzrage | tzanger: lol |
| 17:17.19 | file[laptop] | silly facts - who needs those?!? |
| 17:17.30 | ManxPower | cjk: It's a big hassle and not well supported or documented, but you could always use SIP between Asterisk servers. |
| 17:17.54 | ManxPower | Of course if NAT is involved your head will explode before you get it working. |
| 17:17.57 | cjk | ManxPower, im asking about iax becaus i like some parts of it. plug&play behind nat |
| 17:17.59 | facek_ | ManxPower so what can i use to get a cidname from postgresql database? |
| 17:18.03 | file[laptop] | yay explosion |
| 17:18.28 | ManxPower | facek_: You write an AGI script or write an asterisk_app. |
| 17:18.47 | ManxPower | facek_: There are also at least two scripts to do that mentioned on the mailing list, as well as a service that does that for you. |
| 17:18.49 | blitzrage | wtf! this is really getting annoying. Phone losing registration after a period of time for no reason.... |
| 17:19.00 | blitzrage | well, obviously a reason, I just don't know it :) |
| 17:19.02 | cjk | blitzrage, behind nat this is normal |
| 17:19.02 | ManxPower | blitzrage: phone behind NAT? |
| 17:19.10 | patdk | blitzrage, keep it plugged in |
| 17:19.12 | cjk | blitzrage, tell the to register quite often |
| 17:19.15 | blitzrage | ManxPower: yes, but it used to work fine, and it registers fine. |
| 17:19.26 | facek_ | ManxPower any link? |
| 17:19.26 | luke-jr_ | blitzrage: My * server loses its registration a lot too :/ |
| 17:19.37 | cjk | blitzrage, if they are grandstreams tell them to register every 4 mins |
| 17:19.39 | cjk | this helps |
| 17:19.41 | ManxPower | You can do two things to make that work. qualify=yes or tell your device to register every 60 seconds |
| 17:19.42 | blitzrage | just the phone losing reg... Asterisk works fine |
| 17:19.45 | patdk | hmm, behind nat, you need to set them to atleast every 2min |
| 17:19.53 | blitzrage | cjk: no GS.... 7960 |
| 17:20.04 | ManxPower | facek_: google add site:lists.digium.com to your query |
| 17:20.15 | blitzrage | ManxPower: I'm doing q=yes, thats what I'm trying to debug. Asterisk sends the OPTIONS, no reply from the phone |
| 17:20.27 | cjk | blitzrage, doesnt matter, they should register every few minutes |
| 17:20.34 | patdk | ls |
| 17:20.39 | ManxPower | facek_: but it sounds like you don't want to do any actual research so perhaps paying a consultant to do this for you would be your best option. |
| 17:20.41 | blitzrage | well, I'll worry about that after :) thanks |
| 17:20.42 | facek_ | ManxPower what deafultzone and loadonze should i set in zaptel.conf ? |
| 17:20.56 | blitzrage | facek_: wherever you live |
| 17:21.04 | blitzrage | facek_: if N.A., just leave it default |
| 17:21.08 | _SMP_ | Does anyone know why nufone.net isn't taking registrations? The "system upgrade" statement on the webpage has been there for a while now and it somehow doesn't seem totally kosher. |
| 17:21.16 | ManxPower | blitzrage: have you had qualify=yes for a while or just recently? |
| 17:21.21 | blitzrage | ManxPower: a while |
| 17:21.29 | blitzrage | ManxPower: I had to redo this server, it dided |
| 17:21.47 | ManxPower | blitzrage: weird. set your phone to register every 60 seconds |
| 17:21.55 | blitzrage | ManxPower: I'll give that a shot |
| 17:22.02 | blitzrage | fuck is tcpdump output ever annoying |
| 17:22.25 | file[laptop] | welcome to my life |
| 17:23.30 | blitzrage | is tethereal's output any better for this kind of thing? |
| 17:24.42 | roamer323 | on sip registrations - anyone knows if you there is a way to adjust the registration expiry on a per-registration basis? |
| 17:25.26 | ManxPower | blitzrage: I'm sure it is. tcpdump is pretty primitive |
| 17:25.39 | cjk | whats the difference between type=user and type=friend in iax.conf ? |
| 17:25.53 | JerJer[interop] | a type=user is used to authenticate incoming calls to that asterisk box |
| 17:25.53 | blitzrage | cjk: friend is both user and peer |
| 17:26.09 | JerJer[interop] | a type=friend is both a user and a peer, which is very evil and will bite you someday |
| 17:26.18 | robl^ | and you can bum $50 off a friend, but not a user |
| 17:26.20 | file[laptop] | blitzrage: it's all your fault |
| 17:26.30 | blitzrage | file[laptop]: when is it not? |
| 17:26.59 | file[laptop] | blitzrage: when it's the 32nd day of the month |
| 17:27.00 | cjk | ok so for my iax phones i should use type=friend |
| 17:27.22 | mishehu | grrr. |
| 17:27.30 | facek_ | but MYSQL applicatins exsista, and work good, yes? |
| 17:27.32 | mishehu | rsa auth is still borked on my system... |
| 17:27.35 | *** join/#asterisk FryGuy (~FryGuy@c-24-10-47-136.hsd1.ca.comcast.net) |
| 17:27.56 | mishehu | I can call out to another system, but not call into my system from another one, with rsa. |
| 17:28.13 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E4EF.versanet.de) |
| 17:28.16 | *** part/#asterisk Nebukadneza (~daddel9@i3ED6E4EF.versanet.de) |
| 17:28.21 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E4EF.versanet.de) |
| 17:28.30 | ManxPower | cjk: My general policy is type=friend is for PHONES. type=user/type=peer is for GATEWAYS or SERVERS. |
| 17:28.32 | *** join/#asterisk koolman (~non@70-57-11-107.dnvr.qwest.net) |
| 17:28.51 | cjk | ManxPower, ok thanks that was clear |
| 17:29.24 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 17:29.56 | JerJer[interop] | i still don't like type=friend |
| 17:29.59 | JerJer[interop] | even for phones |
| 17:30.09 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-228-104.dsl.scarlet.be) |
| 17:30.19 | cjk | JerJer[interop], but its the only way it works? at least on my system |
| 17:30.25 | JerJer[interop] | no |
| 17:30.31 | JerJer[interop] | have both |
| 17:30.37 | JerJer[interop] | they can be the same [name] |
| 17:30.57 | cjk | ok so for eaxh phone i create a peer and a user entry?P |
| 17:31.23 | patdk | ya |
| 17:31.28 | JerJer[interop] | correct |
| 17:31.39 | patdk | heh, I do mine like manxpower, friend for phones only |
| 17:31.59 | cjk | sorry JerJer[interop] but then i prefer the type=friend |
| 17:32.23 | JerJer[interop] | trust me, if you ever get a complex asterisk deployment that prefernece will bite you |
| 17:32.50 | roamer323 | when things don't work - type=friend will blow your mind... get a C code debugger/browser ready! |
| 17:33.02 | cjk | JerJer[interop], trust me my * is already quite complex |
| 17:33.15 | JerJer[interop] | not if you are asking these basic questions |
| 17:33.32 | cjk | JerJer[interop], well because i only used sip in the frontend |
| 17:33.40 | cjk | and iax for linking my * |
| 17:33.43 | cjk | and now i offer both |
| 17:36.00 | koolman | Ok, I need help getting this TDM400P configured on my linux box... for some reason the card isn't being fully recogonized. I have setup my copiled zaptel, installed it no issues as I could tell, I setup the udev stuff (this is a FC3 box), Edited my zaptel.conf file, I then rebooted the box, then I ran modprobe zaptel (this seems to go ok) I then run modprobe wctdm and get an error stating "ZT_CHANCONFIG failed on channel 1: No such device or add |
| 17:36.08 | blitzrage | wow... now Asterisk isn't even replying to the REGISTER |
| 17:36.10 | blitzrage | that's fucked |
| 17:36.49 | *** join/#asterisk fsck (~lele@rivendell.windmill.it) |
| 17:36.50 | blitzrage | koolman: reboot after installing udev rules? |
| 17:37.04 | blitzrage | koolman: card plugged into a power connector? (common error, I've done it many times) |
| 17:37.13 | koolman | I checked in /dev/zap/ and from what I've read I need to see device files 1,2,3,4 along with channel, ctl, pseudo, and timer, I see all the device files except for the number device files which should be specifying where the actually FXO card is on my 400P |
| 17:37.15 | koolman | Yes. |
| 17:37.25 | koolman | blitzrage: rebooted after udev |
| 17:37.29 | blitzrage | koolman: paste config to pastebin.ca |
| 17:37.40 | blitzrage | koolman: plus output of what you typed and the error |
| 17:37.48 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 17:38.15 | koolman | blitzrage: card plugged into a power connector? this was my intial issue... then I plugged it in... I still don't see lights but from what I read I shouldn't till the driver gets loaded. |
| 17:38.17 | facek_ | blitzrage i was talikng about that PGSQL http://lists.digium.com/pipermail/asterisk-dev/2003-July/001052.html |
| 17:38.28 | koolman | blitzrage: I will do this.. (past into pastebin.ca |
| 17:38.55 | blitzrage | koolman: no lights until drivers loaded, yes. |
| 17:40.27 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 17:41.11 | koolman | blitzrage: posted in pastebin.ca under my nick koolman |
| 17:41.17 | MuppetMaster | Hi |
| 17:41.23 | blitzrage | koolman: paste the link here |
| 17:41.43 | koolman | Here are my commands |
| 17:41.44 | koolman | http://pastebin.ca/8728 |
| 17:41.47 | mishehu | who can help me debug a problem I'm having with iax and rsa authentication? |
| 17:41.53 | facek_ | what is app_sql_postgres.c ? |
| 17:41.57 | koolman | here is my zaptel.conf http://pastebin.ca/8727 |
| 17:42.19 | mishehu | I can call out of my server to another no problem, but cannot call into my server from another. |
| 17:42.24 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 17:43.41 | mishehu | I've got tcpdump data available. |
| 17:43.46 | blitzrage | koolman: output of uname -a and ls -la /usr/src/linux-2.6 ? |
| 17:44.04 | blitzrage | mishehu: paste it to a pastebin and maybe someone can help |
| 17:44.14 | bkw_ | where is mikej |
| 17:44.21 | mishehu | jbot: pastebin |
| 17:44.22 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 17:44.26 | facek_ | bkw_ what is app_sql_postgres.c |
| 17:44.33 | koolman | blitzrage: http://pastebin.ca/8729 |
| 17:45.12 | blitzrage | is it just me, or is pastebin.ca really slow? |
| 17:45.29 | bkw_ | its app_sql_postgres |
| 17:45.31 | koolman | I've never used it before, but it doesn't seem bad to me.. |
| 17:45.46 | facek_ | bkw_ to make a simpel select to database from extensions, right? how to install that |
| 17:45.54 | bkw_ | why do you ask me? |
| 17:46.01 | cybast | VirTerm => I had my asterisk working and now that I've played around with interrupts etc I can't get the asterisk -vvvc working. I keep getting an ERROR[7747]: mkintf Unabe to open channel 1:No such device or address |
| 17:46.16 | JerJer[interop] | cybast: ztcfg -vvv |
| 17:46.39 | facek_ | bkw_ i like you ;] |
| 17:46.40 | *** join/#asterisk mkhan (~mkhan@dsl092-066-137.bos1.dsl.speakeasy.net) |
| 17:46.46 | cybast | THANKS |
| 17:46.52 | bkw_ | hehe |
| 17:46.55 | cybast | what exactly does that do |
| 17:46.58 | bkw_ | well I have no clue about dat app |
| 17:47.18 | koolman | blitzrage: any ideas? |
| 17:47.31 | mkhan | can anybody help me pls..i have install zaptel, libpri and asterisk.. now.. i did modprobe wtcdm.. getting error.. i think i wil have to modify zaptel.cfg . would anyboyd help pls |
| 17:47.38 | koolman | blitzrage: seems like it should be working.. |
| 17:47.40 | blitzrage | koolman: hrmmm.... I'm pretty stumped..... |
| 17:47.40 | facek_ | bkw_ can you look at it, its in source in asterisdk in apps catalogue. maybe you will know how to install it |
| 17:47.42 | blitzrage | koolman: I agree |
| 17:47.50 | koolman | blitzrage: maybe it's the box |
| 17:48.00 | bkw_ | facek_, not today.. day off.. |
| 17:48.05 | koolman | blitzrage: Do you know any requirements or anything that a PII 450 might not have? |
| 17:48.08 | facek_ | USE_POSTGRES_VM_INTERFACE=0 |
| 17:48.12 | facek_ | hmm.. what that can be |
| 17:48.13 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 17:48.13 | blitzrage | koolman: PCI 2.2? |
| 17:48.21 | bkw_ | facek_, no thats not it |
| 17:48.21 | koolman | blitzrage: how can I tell if it has this? |
| 17:48.24 | blitzrage | koolman: I've been burned on that.... check if its PCI 2.1 |
| 17:48.28 | blitzrage | koolman: have to find a manual |
| 17:48.40 | koolman | blitzrage: hmmmmm damn... |
| 17:48.47 | koolman | so It needs to be PCI 2.1? or 2.2? |
| 17:48.53 | blitzrage | koolman: needs 2.2 |
| 17:48.53 | *** join/#asterisk DannyF (~wizard@c-98f472d5.05-103-73746f40.cust.bredbandsbolaget.se) |
| 17:48.58 | blitzrage | koolman: on a board that old, its probably 2.1 |
| 17:49.01 | koolman | ok... I will look into that.. |
| 17:49.11 | koolman | I have another box I can try so... I will do that.. |
| 17:49.14 | blitzrage | koolman: if its 2.1.... sorry about your luck :) |
| 17:49.15 | cybast | STOP NOW |
| 17:49.25 | blitzrage | koolman: yah, try something PIII or newer |
| 17:49.26 | cybast | wrong terminal . . oops |
| 17:49.42 | koolman | k |
| 17:49.45 | koolman | thanks for the help |
| 17:49.47 | DrukenHME | buy a new board... it's what ? 100 bux? |
| 17:49.50 | blitzrage | koolman: more than likely it'll be 2.2, or you can just verify |
| 17:49.59 | blitzrage | DrukenHME: then you need new RAM, and a new CPU |
| 17:50.10 | DrukenHME | not nessessarily |
| 17:50.27 | *** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net) |
| 17:50.31 | mkhan | can anybody help me pls |
| 17:50.33 | mkhan | can anybody help me pls..i have install zaptel, libpri and asterisk.. now.. i did modprobe wtcdm.. getting error.. i think i wil have to modify zaptel.cfg . would anyboyd help pls |
| 17:50.33 | blitzrage | DrukenHME: good luck on finding a MB that supports PII in a store :) |
| 17:50.48 | JerJer[mobile] | Fry's :) |
| 17:50.58 | blitzrage | mkhan: you have to paste the debug info to a pastebin. That question gives no informatin to help you |
| 17:51.03 | DrukenHME | oh well shit... i wouldn't bother with a PII |
| 17:51.05 | DrukenHME | hehehe |
| 17:51.11 | blitzrage | DrukenHME: :) |
| 17:51.26 | DrukenHME | i have a PII server... it handles... 4 ports? total... |
| 17:51.34 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 17:52.00 | *** join/#asterisk DaLion (DaLion@Toronto-HSE-ppp3881328.sympatico.ca) |
| 17:52.05 | DaLion | hi all |
| 17:52.06 | blitzrage | DrukenHME: he only had 1 |
| 17:52.16 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 17:52.23 | DaLion | anyone could unnderstand this ? |
| 17:52.32 | DaLion | df |
| 17:52.32 | DaLion | Filesystem 1K-blocks Used Available Use% Mounted on |
| 17:52.40 | file[laptop] | use df -h |
| 17:52.43 | mkhan | [root@tuna asterisk]# modprobe wctdm |
| 17:52.44 | mkhan | Notice: Configuration file is /etc/zaptel.conf |
| 17:52.44 | mkhan | line 0: Unable to open master device '/dev/zap/ctl' |
| 17:52.44 | mkhan | 1 error(s) detected |
| 17:52.44 | mkhan | FATAL: Error running install command for wctdm |
| 17:52.44 | DaLion | ./dev/md1 114761296 109376000 0 100% / |
| 17:52.46 | file[laptop] | it'll put it into a human readable format |
| 17:52.58 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 17:52.58 | DrukenHME | ~pastebin |
| 17:52.59 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 17:53.00 | ManxPower | DaLion: use pastebin.ca to paste stuff that's more than 2 or so lines |
| 17:53.01 | koolman | DrukenHME: I have other boxes... just wanted to test things out on this PII450.. it was the best low end box I have.. |
| 17:53.01 | mkhan | and then.. I ran again.. modprobe.. and idnt get any error |
| 17:53.11 | file[laptop] | DaLion: that was directed to you btw |
| 17:53.15 | DaLion | i have space... but it says zero avail.. |
| 17:53.21 | DaLion | i know.. |
| 17:53.29 | JerJer[mobile] | mkhan: have you ran make install on zaptel? |
| 17:53.35 | ManxPower | DaLion: sounds like you have to find a #linux channel |
| 17:53.36 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 17:53.40 | DaLion | so cat /proc/mdstat says its redoing they whole drive.. |
| 17:53.41 | mkhan | yes |
| 17:53.43 | mkhan | [root@tuna asterisk]# lsmod | grep wctdm |
| 17:53.43 | mkhan | wctdm 32832 0 |
| 17:53.43 | mkhan | zaptel 204676 1 wctdm |
| 17:53.50 | DaLion | could it be that while this happening i just done have enough room |
| 17:53.53 | mkhan | does it seems okay ? |
| 17:53.59 | ManxPower | mkhan: put your zaptel.conf on pastebin.ca |
| 17:54.05 | DaLion | man IRC should be threaded |
| 17:54.10 | DaLion | color by thread |
| 17:54.11 | DaLion | ;) |
| 17:54.14 | DaLion | confusing |
| 17:54.17 | cypromis | mkhan: using udev ? |
| 17:54.17 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 17:54.26 | ManxPower | DaLion: no we just have to switch to a web based chat! |
| 17:54.33 | mkhan | cybast, udev?? |
| 17:54.54 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 17:54.59 | cypromis | check README.udev |
| 17:55.08 | DaLion | seems raid drive died or got a severe error.. then it getting rebuild meanwhile .. drive space=0 since i was using 105 out of 110 gig |
| 17:55.10 | JerJer[mobile] | mkhan: are you runnng a 2.6 kernel? |
| 17:55.13 | mkhan | ManxPower, I han't touch zaptel.conf yet |
| 17:55.19 | DaLion | so .. all services down |
| 17:55.19 | mkhan | JerJer[mobile], yes |
| 17:55.20 | JerJer[mobile] | mkhan: that's your problem then |
| 17:55.27 | DaLion | file makes sense ? |
| 17:55.30 | JerJer[mobile] | you have to confgure zaptel.conf for your hardware |
| 17:55.52 | ManxPower | mkhan: Asterisk does not come with magical gnomes to configure it for you. |
| 17:56.02 | mkhan | JerJer[mobile], can u help me to configure it |
| 17:56.07 | ManxPower | ~docs |
| 17:56.08 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:56.15 | mkhan | ManxPower, i know. thats why im here .. :p |
| 17:56.21 | ManxPower | What IS today? "Hold my hand and configure my asterisk for me" day? |
| 17:56.28 | file[laptop] | ManxPower: yes. |
| 17:56.28 | cypromis | seems so |
| 17:56.29 | JerJer[mobile] | mkhan: no but my magical gnome can for $85 an hour |
| 17:56.32 | JerJer[mobile] | his name is shido6 |
| 17:56.33 | facek_ | ha! i have PGSQL: Do several SQLy things |
| 17:56.42 | facek_ | bkw_ i installed PGSQL: Do several SQLy things |
| 17:56.43 | facek_ | ;] |
| 17:56.49 | ManxPower | JerJer[mobile]: I thought shido6 was an Orc! |
| 17:57.02 | mkhan | JerJer[mobile], do you have free trial !! |
| 17:57.06 | blitzrage | I'll do it for $85 CDN :) |
| 17:57.13 | file[laptop] | silly blitzrage |
| 17:57.16 | DaLion | hey mine does for 60 |
| 17:57.17 | DaLion | ;) |
| 17:57.19 | blitzrage | kram: ahoi! |
| 17:57.22 | JerJer[interop] | mkhan: not for configuration |
| 17:57.24 | kram | greets blitz |
| 17:57.29 | ManxPower | I'lll do it for free for a serious job offer. |
| 17:57.31 | DaLion | hes called.. h,mm.. well .. `hostname` |
| 17:57.46 | mkhan | JerJer[interop], lol.. thx.. i am not in business.. do this as hobby .. u can say |
| 17:57.49 | shido6 | Orc? |
| 17:57.51 | shido6 | me? |
| 17:57.53 | JerJer[interop] | plus there is nothing in this world that is 'without cost' |
| 17:57.59 | JerJer[interop] | mkhan: famous last words |
| 17:58.00 | *** join/#asterisk Blackvel (~blackvel@dsl-082-083-169-239.arcor-ip.net) |
| 17:58.07 | DrukenHME | hey bkw_, how's the girlfriend (allison)? |
| 17:58.10 | DaLion | free |
| 17:58.16 | DaLion | doesnt exist ? hooh man |
| 17:58.18 | shido6 | thats how NuFone started... actually |
| 17:58.21 | shido6 | as a hobby |
| 17:58.34 | JerJer[interop] | yeah kinda |
| 17:58.44 | mkhan | can anybody help me.. without asking for any USD, CAN $ ? |
| 17:58.47 | ManxPower | shido6: Well EVERYONE wants to be a 5th class Wizard. |
| 17:58.50 | bkw_ | DrukenHME, she's great .. haha |
| 17:58.54 | JerJer[interop] | plus nobody was doing IAX termination at that time |
| 17:59.01 | DrukenHME | bkw_: :) |
| 17:59.12 | ManxPower | mkhan: I'm sure there is someone weird enough to do so, but we all had to read the docs and the mailing list archives |
| 17:59.19 | blitzrage | ok... so.... why would tcpdump see a REGISTER< but Asterisk not pick it up.... |
| 17:59.31 | DaLion | mkhan sure |
| 17:59.45 | ManxPower | blitzrage: all the ip addresses of the server are in /etc/hosts? |
| 17:59.51 | mkhan | DaLion .. would u help me.. ? |
| 18:00.10 | mishehu | http://pastebin.ca/8730 for anybody willing to help me with my RSA problem |
| 18:00.18 | ManxPower | blitzrage: chan_sip can get a little weird if it can't resolve the addresses of the box. |
| 18:00.42 | blitzrage | hrmmmmmmmmmmmmmmmmmmmm, interesting |
| 18:00.46 | blitzrage | ManxPower: I'll give that a shot |
| 18:00.53 | mkhan | DaLion, i am in zaptel.conf.. what should I choose.. span or dynamic? |
| 18:01.08 | blitzrage | hrm, seems to resolve itself fine |
| 18:01.30 | ManxPower | blitzrage: what are you using to test resolution? |
| 18:01.31 | DaLion | hey |
| 18:01.41 | blitzrage | ManxPower: packet sniffs on both ends |
| 18:01.52 | DaLion | not sure |
| 18:01.54 | blitzrage | ManxPower: and sip debug ip <blah> |
| 18:02.08 | ManxPower | blitzrage: that doesn't test address -> name resolution |
| 18:02.16 | *** part/#asterisk DaLion (DaLion@Toronto-HSE-ppp3881328.sympatico.ca) |
| 18:02.22 | blitzrage | oh, I just did a ping |
| 18:02.32 | ManxPower | blitzrage: that doesn't test resolution either |
| 18:02.34 | blitzrage | ping <FQDN> |
| 18:02.39 | blitzrage | ManxPower: ok... how then? :) |
| 18:02.42 | ManxPower | try host 1.2.3.4 |
| 18:02.51 | ManxPower | or whatever the address of the server is. |
| 18:03.05 | blitzrage | yep, worked |
| 18:03.32 | ManxPower | blitzrage: You are not doing something stupid like portforwarding on the NAT box, are you? |
| 18:04.06 | blitzrage | ManxPower: the addresses are being forwarded to the phone on the NAT box, Asterisk is on an external IP with port 5060 open in both directions. |
| 18:04.23 | blitzrage | sip.conf has nat=yes for the phone |
| 18:04.31 | ManxPower | blitzrage: don't port forward on the NAT router for the phone. |
| 18:04.54 | ManxPower | blitzrage: does the asterisk server have a private address too? |
| 18:04.58 | *** join/#asterisk junbug (junya@adsl-3-237-168.mia.bellsouth.net) |
| 18:05.16 | blitzrage | Asterisk is on an external IP |
| 18:05.25 | ManxPower | blitzrage: what do you mean by "Asterisk is on an external IP with port 5060 open in both directions" |
| 18:05.42 | ManxPower | blitzrage: If the asterisk server has a public and a private IP it might be using the private IP |
| 18:05.49 | *** part/#asterisk lbarth (user@pD9EA607A.dip.t-dialin.net) |
| 18:05.56 | blitzrage | ManxPower: packet sniffs don't show that... |
| 18:06.06 | blitzrage | sees the registration, does nothing about it |
| 18:06.13 | blitzrage | maybe I need to restart Asterisk... |
| 18:06.15 | mishehu | hrmf. |
| 18:06.18 | *** join/#asterisk MPreuett (~mpreuett@pcp03933704pcs.sthind01.mo.comcast.net) |
| 18:06.19 | ManxPower | blitzrage: Trust me. |
| 18:06.34 | MPreuett | greetings everyone. |
| 18:06.53 | ManxPower | blitzrage: "netstat -an | grep 5060" |
| 18:07.17 | ManxPower | MPreuett: I'm sorry, but not many will configure Asterisk for you for free." |
| 18:07.26 | blitzrage | udp 0 0 0.0.0.0:5060 0.0.0.0:* |
| 18:07.37 | *** join/#asterisk imagmo (~imagmo@c-24-20-249-117.hsd1.or.comcast.net) |
| 18:08.00 | ManxPower | blitzrage: make sure "iptables -L" and "iptables -L -t nat" are empty |
| 18:08.41 | blitzrage | nat is empty, -L is not because there is a firewall on |
| 18:08.58 | ManxPower | blitzrage: sounds like you need to turn off the firewall for a while. |
| 18:09.08 | blitzrage | ManxPower: I think the INPUT chain is wrong.... |
| 18:10.03 | blitzrage | 0 0 ACCEPT udp -- eth0 any anywhere anywhere udp spt:5060 dpt:5060 |
| 18:10.30 | blitzrage | actualy... I don't think its wrong... but I think its wrong because its not matching anything |
| 18:10.41 | ManxPower | blitzrage: what makes you think the SOURCE port is going to be 5060 from the SIP client (ESPECIALLY with a NAT'd client)? |
| 18:11.08 | blitzrage | packet traces all seem to be 5060 |
| 18:11.30 | blitzrage | let me get rid of SPT |
| 18:11.38 | ManxPower | blitzrage: Your problem is weird enough I would not trust anything. Put a match and log everything at the end of your firewall rules or just turn it off for testing. |
| 18:16.32 | blitzrage | god I need to get this figured out soon so I can go and make breakfast |
| 18:16.43 | cybast | anyone know where the feature activation codes for thing like dnd reside (ie what conf file) |
| 18:16.56 | ManxPower | cybast: 1.0.x or CVS-HEAD? |
| 18:17.07 | cybast | CVS-HEAD |
| 18:17.30 | ManxPower | cybast: features.conf but I think DND and stuff like that is only handled in chan_zap |
| 18:17.50 | cypromis | .w 20 |
| 18:18.08 | cybast | I looked in features.conf and it wasn't there |
| 18:18.17 | blitzrage | configure DND in dialplan logic |
| 18:18.21 | blitzrage | thats what I do |
| 18:18.32 | cybast | is there a reference to all the feature activation codes somewhere |
| 18:18.33 | ManxPower | cybast: then it's prolly handles only in chan_zap or do what blitzrage said. |
| 18:19.02 | blitzrage | just save the status to AstDB, and check it before you place a call to the extension |
| 18:19.27 | cybast | sorry, I'm an asterisk newbie where is the dialplan |
| 18:19.34 | mishehu | *sigh* |
| 18:19.35 | blitzrage | extensions.conf |
| 18:19.41 | blitzrage | cybast: good luck :) |
| 18:19.46 | cybast | got ya . .thanks |
| 18:19.54 | mishehu | I don't understand why RSA auth stopped working for me. :-/ |
| 18:20.36 | ManxPower | mishehu: using type=friend? |
| 18:20.58 | mishehu | ManxPower: yeah, as I need both inbound and outbound |
| 18:21.08 | mishehu | was tehre a change to how these worked in recent times? |
| 18:21.12 | ManxPower | mishehu: one side is not sending the RSA keys. |
| 18:21.41 | mishehu | ManxPower: one side is definitely sending an IAX2 INVAL even before sending the AUTHREP with the RSA key |
| 18:21.46 | ManxPower | mishehu: well a couple of bugs fixed with clienting being able to connect and not be authorized. |
| 18:21.46 | mishehu | you are correct about that. |
| 18:21.56 | ManxPower | mishehu: why not use secret=? |
| 18:22.03 | VirTERM | cybast: did you get it? |
| 18:22.30 | mishehu | ManxPower: would breaking it up into two entries, one of type user, one of type peer, possibly resolve the RSA key issue? |
| 18:22.48 | ManxPower | mishehu: it could. |
| 18:23.04 | ManxPower | Since using type=friend for SERVERS really WILL bite you eventually. |
| 18:23.16 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 18:23.24 | *** join/#asterisk topping (~topping@cpe-24-210-82-196.columbus.res.rr.com) |
| 18:23.31 | shido6 | Im not gonna say anything... |
| 18:23.32 | mishehu | ManxPower: I'll try it. I have to go eat something now though... |
| 18:23.33 | ManxPower | mishehu: PASTE the Dial line., |
| 18:23.38 | *** join/#asterisk Duy (~duy@port-83-236-189-65.static.qsc.de) |
| 18:23.48 | mishehu | ManxPower: http://pastebin.ca/8730 |
| 18:24.19 | ManxPower | mishehu: Dialing by IP or hostname will make Asterisk ignore the settings in anything except [general] |
| 18:24.20 | mishehu | but nonetheless, if for a server type friend will bite me in the ass, as it might already be doing, I'll fix that too. |
| 18:24.50 | Duy | Hello, can someone help me! I got always a fake ringtone during the connection Time, how can I disable it? |
| 18:25.05 | mishehu | ManxPower: oh shit. it should be Dial(IAX2/rakdanit/s@mainmenu) right? |
| 18:25.15 | ManxPower | Duy: remove the Fake Ring Tone Option. ("r" on the Dial line) |
| 18:25.26 | Duy | oh ok thx manx power |
| 18:25.37 | TechDawg | I'm having trouble compiling zaptel. I get this error: /usr/include/linux/module.h:21: linux/modversions.h: No such file or directory |
| 18:25.46 | TechDawg | I know I'm missing something but don't know what. |
| 18:25.48 | ManxPower | mishehu: I usually use Dial(IAX2/username@iaxconfentry/extension) |
| 18:26.00 | ManxPower | put the secret= in the [isxconfentry] |
| 18:26.42 | ManxPower | You don't normally need the @context. |
| 18:27.02 | ManxPower | TechDawg: you are missing glibc-headers or something like that. |
| 18:27.16 | TechDawg | That's a start, let me check. |
| 18:27.25 | ManxPower | on my system that file is in glibc-headers |
| 18:27.41 | ManxPower | sorry in glibc-devel |
| 18:27.44 | ManxPower | [eric@vulcan eric]$ rpm -qf /usr/include/linux/module.h |
| 18:27.44 | JerJer[interop] | it is the kernel headers you are missing |
| 18:27.44 | ManxPower | glibc-devel-2.3.3-23.1.101mdk |
| 18:28.36 | mishehu | ManxPower: I'll try your suggestions soon as I eat lunch |
| 18:28.40 | mishehu | thanks for the help |
| 18:28.56 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 18:30.43 | *** join/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 18:31.36 | blitzrage | ManxPower: working now. Not sure what I was thinking about the source port... I think that was the problem. |
| 18:31.49 | blitzrage | ManxPower: you're right, I don't know what I assume the source port should be 5060. |
| 18:31.56 | blitzrage | s/what/why |
| 18:31.58 | ManxPower | blitzrage: Even I sometimes forget that the source port is usually random on the client side. |
| 18:32.10 | blitzrage | ManxPower: yeppers |
| 18:32.26 | ManxPower | blitzrage: so, when are you going to start putting my stuff on asteriskdocs.org? 8-) |
| 18:32.37 | blitzrage | ManxPower: I have no idea how to put it on there.... |
| 18:32.43 | blitzrage | ManxPower: stupid Xoops |
| 18:32.46 | *** part/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 18:32.57 | ManxPower | blitzrage: people keep asking me for the site. |
| 18:32.59 | blitzrage | ManxPower: I can't seem to figure out a good place for it... |
| 18:33.02 | blitzrage | ManxPower: I know, me too... |
| 18:33.13 | ManxPower | blitzrage: Yeah, that can be an issue. |
| 18:33.26 | ManxPower | perhaps a config examples section for the example config files? |
| 18:34.16 | blitzrage | ManxPower: problem is... if I make a new box on the left side for a "Contributed Info" or something liek that, then I'm going to have to write a custom page for all the links, then after that, I don't know how to link them in Xoops... |
| 18:34.35 | blitzrage | ManxPower: like, look at the site, and tell me where you think it should go? |
| 18:34.43 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:34.56 | blitzrage | I;ve looked at it a hundred times, everytime I go to try and add it, I get lost as to where it should go... |
| 18:35.23 | ManxPower | blitzrage: Would it be better if you gave the info to someone more familiar with the software used to manage the docs? |
| 18:35.38 | blitzrage | ManxPower: that's me :) |
| 18:35.49 | ManxPower | Configuring Channels |
| 18:35.49 | ManxPower | <PROTECTED> |
| 18:35.49 | ManxPower | <PROTECTED> |
| 18:35.50 | blitzrage | ManxPower: I'm the only one who uses the site (admin wise) |
| 18:35.54 | ManxPower | might be a food place |
| 18:36.02 | ManxPower | food == good. |
| 18:36.45 | ManxPower | and of course my extensions.conf sample can go into the dialplans section |
| 18:37.03 | JerJer[interop] | so SIP RTP |
| 18:37.27 | JerJer[interop] | does anyone know if there is anything in the spec that defines how RTP starts |
| 18:37.32 | *** join/#asterisk PBXtech (~nik@wirelessdata-167-246.mycingular.net) |
| 18:37.39 | JerJer[interop] | like who sends the RTP first? |
| 18:37.47 | file[laptop] | depends |
| 18:38.04 | file[laptop] | cause you can have an RTP stream occur before the actual call is up, ie: inband progress... |
| 18:38.04 | *** join/#asterisk __Crash (~me@195.158.83.189) |
| 18:38.06 | JerJer[interop] | i figured it was who initiates the call (ie early media) |
| 18:38.49 | file[laptop] | that would be interesting to test |
| 18:39.00 | harryvv | Jer do you know if a end customer needs a min of 4 rtp ports open? Somone I was assisting last night still could not hear two way audio even after opening up 10001-10004. I have 10001-10010 open. Or mabey his windows xp firewall mabey causing problems. He can access my voicemail as I was watching on the cli. |
| 18:39.12 | smurfix | JerJer[interop]: Why should the call initiator send anything when the call's not set up yet? |
| 18:39.23 | file[laptop] | smurfix: that's what I was thinking |
| 18:40.10 | __Crash | Is there some standard documentation supporting how to connect a SIP client to an Asterisk running behind a firewall> |
| 18:40.30 | JerJer[interop] | file[laptop]: that is what we are pondering now |
| 18:40.33 | smurfix | It's bidirectional UDP anyway, so I suppose you send RDP packets as soon as you (a) know where to send them and (b) have something to send, regardless of what the other side does with their audio |
| 18:40.47 | harryvv | Crash just 5060 and rtp ports. If your running stun then I think those also need to be opened. |
| 18:40.58 | Duy | ManxPower: as I remove the r out of the Dial Line, I hear no Ringtone on my phone although the other side Handy rings |
| 18:41.02 | blitzrage | fuck fuck fuck |
| 18:41.09 | blitzrage | you know when you you're just having one of those days.... that nothing works |
| 18:41.10 | JerJer[interop] | so how do we figure if there is early media"? |
| 18:41.31 | file[laptop] | JerJer[interop]: you get SDP in a sip reply? |
| 18:41.36 | blitzrage | makes no sense... SIP poking works fine, then I place a call, then it breaks |
| 18:41.50 | harryvv | you get audio then it breaks? |
| 18:41.58 | blitzrage | no |
| 18:42.05 | harryvv | or the session |
| 18:42.48 | file[laptop] | JerJer[interop]: usually for early media though you'll get a 183 Session Progress with SDP |
| 18:42.58 | blitzrage | qualify works for a period of time, I place a call, works, hang up, then qualify no longer gets to the phone, asterisk keeps retransmitting the OPTIONS, then calls don't work. |
| 18:43.05 | file[laptop] | JerJer[interop]: but I was mucking around once with 180 Ringing and SDP, and the gateway still used the SDP and RTP stream for audio |
| 18:43.40 | Duy | Someone can held me? As I remove the r Option for fake Rington (the Dial Line), I hear no Ringtone on my phone although the other side of the line rings |
| 18:43.46 | Zeeek | blitzrage are you using multiple clients behind NAT? |
| 18:43.56 | blitzrage | Zeeek: yes, but only one registers |
| 18:44.08 | Zeeek | I have had the problems you describe even with IAX |
| 18:44.21 | Zeeek | (well the ons I've seen in the last few lines) |
| 18:44.45 | Zeeek | I have multiple SIP clients working though |
| 18:45.19 | ManxPower | Duy: Asterisk will provide ring tone if it thinks it should. I don't know why it doesn't think it should. |
| 18:45.21 | Zeeek | client2 set to 5061with 5061 forwarded to its ip |
| 18:45.34 | Zeeek | and a different RTP range |
| 18:45.54 | Duy | ManxPower ok |
| 18:46.05 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-196-171.dsl.scarlet.be) |
| 18:46.23 | ManxPower | Duy: What is your ACTUAL Dial line (sans password)? |
| 18:46.34 | |Vulture| | Anyone ever use the QoS DiffServ on a Netgear Layer2 Switch? |
| 18:47.45 | harryvv | Zeek why 5061 |
| 18:47.58 | Zeeek | I liked the sound of it |
| 18:48.08 | harryvv | ookay |
| 18:48.16 | Zeeek | and it works flawlessly |
| 18:48.30 | Zeeek | actually I believe someone on the FWD forums mentioned this |
| 18:48.35 | *** join/#asterisk Corydon76-home (ten@pcp08665860pcs.500ash01.tn.comcast.net) |
| 18:49.02 | Zeeek | <PROTECTED> |
| 18:50.09 | ManxPower | Zeeek: The guy forgot localnet= and nat=yes |
| 18:50.24 | harryvv | btw what is the purpous of rtf having a wide range of ports open? to accomidate several voice sessions? |
| 18:50.30 | Zeeek | who? willy? |
| 18:51.04 | Zeeek | I think he guessed that people would have read the docs :) |
| 18:51.20 | Duy | ManxPower: my dial plan is all call trough exten => _0.,1,Dial(SIP/${EXTEN:1}@CC,,t) |
| 18:51.21 | Duy | exten => _0.,2,Hangup |
| 18:52.27 | ManxPower | Duy: does the call actually work other than the ring sound problem? |
| 18:53.30 | JerJer[interop] | file[laptop]: so if there is no early media, how does the RTP start? |
| 18:53.42 | Duy | ManxPower: Duy the call actually work I can phone and talk I have only the ring sound problem, with the option r I got a fake ringtone what I not want, without r I got no ringtone but the otherside Telefone ring |
| 18:54.05 | JerJer[interop] | or who starts first? |
| 18:54.39 | file[laptop] | JerJer[interop]: they both start around the same time I'd say... |
| 18:54.42 | ManxPower | Duy: ring sound problems are very hard to diagnose. That's why the "r" option is so popular. Most people consider it too much work to actually fix the issue. |
| 18:55.04 | JerJer[interop] | file[laptop]: ok |
| 18:55.18 | ManxPower | JerJer[interop] wrote chan_skinny (which uses RTP), shouldn't he already know this obsecure stuff? |
| 18:55.34 | file[laptop] | JerJer[interop]: when the 200 OK occurs from the remote side it has the SDP data for the RTP stream for the remote side and that's when the call is answered |
| 18:55.35 | JerJer[interop] | these are questions we are posing for interop |
| 18:55.40 | ManxPower | JerJer[interop]: I'll bet the RTP streams are started independently. |
| 18:55.51 | JerJer[interop] | and we want others input before we test |
| 18:56.06 | JerJer[interop] | gota switch ssids - might die |
| 18:56.08 | file[laptop] | that's on a 180 Ringing when you're getting progression out of band |
| 18:56.23 | file[laptop] | when it's a 183 Session Progress the SDP data is contained in there, and the RTP stream is started then for the remote side... |
| 18:56.36 | file[laptop] | as for the orginating point I'd say it's audio is used when the call is answered at the 200 OK |
| 18:56.43 | JerJer[interop] | ahh ok |
| 18:56.44 | bkw_ | ok why do companies even bother putting stuff on the web if they don't have a way for you to actually buy it |
| 18:56.45 | JerJer[interop] | brb |
| 18:56.46 | bkw_ | its pointless |
| 18:56.48 | __Crash | Does anyone know of a multi-line IAX hard phone??? |
| 18:56.57 | file[laptop] | enough of my ranting now |
| 18:56.59 | shido6 | bkw - whats up? |
| 18:57.20 | *** join/#asterisk vaewynAFK (freeman@mail.deltamach.com) |
| 18:57.21 | Qwell | bkw_: They expect you to click the mailto:sales@mycompany.com link |
| 18:57.22 | shido6 | the real question is |
| 18:57.26 | shido6 | what are you looking to buy |
| 18:57.32 | Qwell | I always just go to another site when I see that |
| 18:57.38 | file[laptop] | he's looking for an electronic dildo! |
| 18:59.00 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 19:00.45 | bjohnson | __Crash: no |
| 19:02.06 | blitzrage | anyone know the command for the SIP<mac>.cnf for a Cisco phone to set the registration timeout? |
| 19:02.15 | blitzrage | ManxPower: I'm looking in your direction :) |
| 19:02.30 | ManxPower | blitzrage: I only use polycoms now |
| 19:03.10 | darkskiez | # Phone Registration Expiration [1-3932100 sec] (Default - 3600) |
| 19:03.10 | darkskiez | timer_register_expires: 3600 |
| 19:03.13 | darkskiez | that? |
| 19:03.25 | blitzrage | darkskiez: thanks! I'll give that a shot |
| 19:03.37 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 19:04.17 | blitzrage | no quotes around the 3600 eh? |
| 19:04.19 | darkskiez | What is remote party-id used for in sip? I've looked it up, but i dont getit. |
| 19:04.20 | nesys | Hi folks ... I've a problem with * and ccme (* as voicemail) ... there's someone that uses * as ccme voicemail? |
| 19:04.40 | darkskiez | blitzrage: no |
| 19:04.46 | blitzrage | okie, thanks |
| 19:05.06 | blitzrage | maybe thats the problem with my start and end_media_port lines too |
| 19:05.24 | *** join/#asterisk JerJer[mobile] (1000@dhcp-11-147.hstf.interop.net) |
| 19:06.38 | bjohnson | do you guys think vlc might work as a mp3 player for MOH. It sounds like it could play a stream such as one from slimserver: http://software.newsforge.com/article.pl?sid=05/03/23/1911207&from=rss |
| 19:07.29 | blitzrage | hrmmmm, wish you could enable NAT on the phone on a line by line basis |
| 19:07.43 | bjohnson | rawplayer? |
| 19:08.00 | blitzrage | /usr/src/asterisk/contrib/utils/ |
| 19:08.01 | darkskiez | blitzrage: i wish you could set speeddials on the phone |
| 19:08.06 | ManxPower | blitzrage: nat=yes means you don't have to enable NAT on the phone |
| 19:08.22 | blitzrage | ManxPower: right! |
| 19:08.35 | blitzrage | hey, everyone send ManxPower a $1... he's smart! :) |
| 19:09.00 | darkskiez | I dont quite get the use of multiple lines, what does it let you do? |
| 19:09.12 | blitzrage | darkskiez: register multiple lines to different asterisk boxes? |
| 19:09.17 | blitzrage | darkskiez: thats what I'm doing anyways. |
| 19:10.02 | darkskiez | but multiple line appearances on one box, maybe for ease of selection of outgoing callerids, but I cant see the massive use. |
| 19:10.28 | ManxPower | My 4-line phone - Line 1: business exten, Line 2: personal extension, Line 3: lover #1 extension (rings at the same time as their home extension), Line 4: lover #2 extension (ings at the same time as their home extension) |
| 19:10.48 | JerJer[interop] | ok so the premise is that the process of RTP is not deterministic |
| 19:10.49 | bjohnson | eventually with better indication support in future hardware versions .. it might lead to line in use type indications like on key systems |
| 19:10.56 | blitzrage | lol, love it. Lover #1 and Lover #2. |
| 19:11.05 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 19:11.20 | ManxPower | or on my customer's phones: Line 1: personal extension, Line 2: Main business number, Line 3: Main business number |
| 19:11.55 | JerJer[interop] | meaning there is more than a single way it can happen |
| 19:12.39 | darkskiez | i wish there was a way of making the phone show you the name of the number you were calling, reverse callerid type thing, like on a mobile phone. |
| 19:12.50 | ManxPower | MY users CANNOT get the hang of call waiting on the analog phones, even though it works EXACTLY like their home call waiting service. |
| 19:12.55 | bjohnson | blitzrage: have you used rawplayer with a stream source? like slimserver? |
| 19:13.06 | ManxPower | darkskiez: Polycom does that if the number is in the directory |
| 19:13.16 | bjohnson | it seems to only play .raw files .. not streams |
| 19:13.41 | darkskiez | ManxPower: I see |
| 19:14.23 | ManxPower | Come to think of it, SIP-841 does that if the number is in the callerid history of incoming calls. |
| 19:14.23 | *** join/#asterisk Gronker__ (~Gronker2@adsl-220-64-104.ags.bellsouth.net) |
| 19:15.08 | blitzrage | bjohnson: nope, just local files |
| 19:15.31 | darkskiez | Thats interesting |
| 19:16.32 | ManxPower | I seem to vaguely recall that some people are using madplay for MoH |
| 19:16.42 | blitzrage | :) |
| 19:17.02 | bjohnson | I guess my inquiry about using vlc to play streams for moh still stands |
| 19:17.18 | bjohnson | I couldn't get madplay to play a stream |
| 19:17.20 | bjohnson | just files |
| 19:18.08 | *** join/#asterisk ___Crash (~me@195.158.83.189) |
| 19:18.21 | bjohnson | mpg123 is the only thing I;ve been able to get to play a stream so far .. but it keeps dying out. A better solution "should" be available. I played with mplayer a bit and it looked promising but I didn't actually get it to work |
| 19:18.23 | ManxPower | I don't suppose anyone knows a command to get the current resolution of the X server? |
| 19:18.38 | bjohnson | look for it in the xsessions file? |
| 19:18.57 | tzanger | ManxPower: xdpyinfo |
| 19:19.40 | tzanger | $ xdpyinfo -display :0 | grep dimensions |
| 19:19.40 | tzanger | <PROTECTED> |
| 19:20.12 | blitzrage | ahhhh, much better |
| 19:20.12 | ManxPower | tzafrir: that's what I was looking for, |
| 19:20.45 | ManxPower | I just wish I could get the damn thing to run in 1152xmumble |
| 19:21.18 | ManxPower | bbiaw |
| 19:24.48 | JerJer[interop] | ok now lets talk about message waiting indication |
| 19:25.00 | JerJer[interop] | does asterisk deal with the subcribe method of MWI? |
| 19:25.13 | blitzrage | thats a good question |
| 19:25.41 | *** part/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 19:26.08 | bjohnson | no idea what that even means |
| 19:26.55 | tzanger | http://64.236.34.67:80/stream/1003 some good trance on now |
| 19:27.03 | tzanger | more ambient than trance really |
| 19:27.45 | blitzrage | tzanger: I prefer Groove Salad |
| 19:27.58 | tzanger | bjohnson: basically subscribe method of MWI is when the phone says "Asstrick dude... I want the 411 on that 500 shizzle, yo" |
| 19:28.11 | blitzrage | lol |
| 19:28.30 | tzanger | and asterisk will mark the phone ofr updates wheneve the state of box 500 changes |
| 19:28.37 | L|NUX | can some one help me in setting up SIP to one line PSTN setup :$ |
| 19:28.42 | L|NUX | or any good link ? |
| 19:28.45 | tzanger | but it'll probably purposely wait 5 minutes beofre telling the phone because the phone called it 'asstrick' |
| 19:29.01 | Nugget | http://slacker.com/photos/lc0405/IMG_3873 <-- vroom |
| 19:29.29 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 19:29.53 | tzanger | you've got strange taste in vroom |
| 19:29.58 | tzanger | http://nciv.flabber.nl/index.php?nciv_id=722246 <-- now THAT's vroom |
| 19:30.26 | tzanger | nice car though |
| 19:31.14 | tzanger | you know what's actually disgusting |
| 19:31.27 | tzanger | I just noticed "terri shiavo autopsiephoto" on there and followed it |
| 19:31.29 | tzanger | good lord |
| 19:31.30 | blitzrage | tzanger: vroom! :) |
| 19:31.32 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 19:32.56 | *** join/#asterisk eivind (~eivindtr@062016241059.customer.alfanett.no) |
| 19:33.48 | JerJer[interop] | from what I can see we just get a message from asterisk saying "bitch you have 5 unread messages" |
| 19:33.55 | L|NUX | any one :( |
| 19:34.12 | tzanger | L|NUX: you need to do some *basic* research |
| 19:34.21 | tzanger | this kind of hand-holding I charge $175/hr for |
| 19:34.49 | tzanger | ahh |
| 19:34.53 | tzanger | that's not terri schiavo |
| 19:35.00 | tzanger | that's Lisa McPherson |
| 19:35.02 | blitzrage | tzanger: I'd hope not |
| 19:35.14 | tzanger | I was wondering why she looked so bad |
| 19:35.21 | blitzrage | oh burn |
| 19:36.28 | JerJer[interop] | ok how about do not disturb - is there anything in the SIP protocol that deals with that? |
| 19:37.33 | tzanger | JerJer[interop]: wouldn't that just be handled within *? |
| 19:37.56 | JerJer[interop] | or the phone itself - i'm trying to determine that |
| 19:38.46 | *** join/#asterisk caesar2 (caesar@p5497EF2B.dip.t-dialin.net) |
| 19:38.47 | PTG1234 | Anyone here have a donotcall.gov account? |
| 19:39.09 | blitzrage | JerJer[interop]: I haven't seen anything in the SIP spec in regards to DND |
| 19:39.21 | shido6 | so what happens when you set the crisco to DND |
| 19:39.36 | blitzrage | shido6: interesting question :) |
| 19:39.55 | blitzrage | lets see! |
| 19:39.57 | tzanger | I can't find the button on the block of lard |
| 19:40.08 | shido6 | <PROTECTED> |
| 19:40.14 | L|NUX | tzanger : hmm |
| 19:40.21 | shido6 | i get 486 back when I set my crisco to DND |
| 19:40.22 | L|NUX | i setup sip to sip right now |
| 19:41.10 | blitzrage | shido6: ahhhhh yes.... that makes sense |
| 19:41.12 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
| 19:41.26 | JerJer[interop] | ok so the phone just tells the proxy "hey i'm busy, go away" |
| 19:41.26 | shido6 | SIP/2.0 486 Busy here |
| 19:41.37 | ariel_ | L|NUX, how is your setup? if your going from sip to pstn on the same system it should be easy via a context and a dialing rule. |
| 19:41.37 | JerJer[interop] | good |
| 19:41.59 | JerJer[interop] | ok - call forwarding |
| 19:42.20 | L|NUX | ariel_ : can you give me example ? |
| 19:42.53 | ariel_ | exten => X.,1,Dial(Zap/1/${EXTEN}) |
| 19:43.33 | blitzrage | ariel_: you forgot the _ |
| 19:43.34 | L|NUX | can i pvt with you ? |
| 19:44.08 | ariel_ | exten => _X.,1,Dial(Zap/1/${EXTEN}) yes your right blitzrage |
| 19:45.12 | *** join/#asterisk deRost (~deRost@054.209-89-66-0.interbaun.com) |
| 19:45.42 | *** join/#asterisk VirTERM (~VirTERM@204.225.113.90) |
| 19:46.11 | *** part/#asterisk VirTERM (~VirTERM@204.225.113.90) |
| 19:46.18 | *** join/#asterisk VirTERM (~VirTERM@204.225.113.90) |
| 19:47.06 | *** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com) |
| 19:48.57 | JerJer[interop] | i don't believe asterisk has any SIP protocol specific call fowarding implemenation |
| 19:49.08 | JerJer[interop] | anyone else see anything different? |
| 19:50.18 | ariel_ | JerJer[interop], no it does not. |
| 19:50.39 | JerJer[interop] | ok good |
| 19:50.47 | JerJer[interop] | well not good - but good for now :) |
| 19:51.52 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 19:52.44 | ariel_ | I wish that the incominglimit and outgoinglimit was back in and fixed correctly the setgroup and groupcount sucks to get working right. In fact it works less then the incominglimit did. |
| 19:52.51 | *** join/#asterisk cia (~cwj@adsl-68-77-11-148.dsl.emhril.ameritech.net) |
| 19:53.04 | *** join/#asterisk Tili (~Tili@202-133-65-163-dialup.sat.net.pk) |
| 19:54.34 | mishehu | hmm... on server 1, I have in iax.conf a peer, with a username provided. on server 2, I have a user, with the same username provided. when I call from server 1 to server 2, I get "no authority". I am trying to use rsa keys, and dialing like exten => 1,Dial(IAX2/thecontext/s@destination_context) |
| 19:54.51 | JerJer[interop] | !? |
| 19:54.56 | JerJer[interop] | mishehu: that makes no sense |
| 19:55.16 | JerJer[interop] | Dial,IAX2/username@peer/exten |
| 19:55.20 | JerJer[interop] | <PROTECTED> |
| 19:55.48 | JerJer[interop] | if your system is setup 'correctly' you shouldn't care what the remote context is |
| 19:57.35 | mishehu | sec, let me pastebin this |
| 20:00.21 | JerJer[interop] | then do you have auth=rsa and outkey defned? |
| 20:02.43 | JerJer[interop] | ok - call hold ... any sip protocol specific functions |
| 20:02.54 | hardwire | blah |
| 20:02.56 | JerJer[interop] | looks like just set rtp 0.0.0.0 |
| 20:03.30 | mishehu | http://pastebin.ca/8737 |
| 20:04.37 | mishehu | hopefully nice and readable. |
| 20:04.50 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 20:05.45 | JerJer[interop] | and then it looks like a=sendonly |
| 20:06.13 | file[laptop] | JerJer[interop]: yeah call hold is just a reinvite to 0.0.0.0 yay |
| 20:06.16 | file[laptop] | easy to identify |
| 20:06.25 | JerJer[interop] | but it looks like that method is depreciated |
| 20:06.33 | file[laptop] | everything I've found uses that |
| 20:06.37 | JerJer[interop] | >1. use the way in RFC3264 since this is newer. > The sendonly, recvonly, inactive attributes give > a little more control than the old "0.0.0.0" method. > |
| 20:06.46 | file[laptop] | freaky though |
| 20:06.54 | file[laptop] | SIP and SDP changes so much |
| 20:07.00 | JerJer[interop] | >But also: >2. If someone sends you a re-INVITE with "0.0.0.0" > try to accept it as described in RFC2543 so that > you are backward-compatible with implementations > that use it. |
| 20:11.32 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 20:15.24 | *** join/#asterisk ckruetze (~nospam@i3ED65B54.versanet.de) |
| 20:18.54 | *** join/#asterisk adker (~adker@67-51-237-86.dsl1.glv.ny.frontiernet.net) |
| 20:22.23 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:22.42 | tzanger | hahah |
| 20:22.43 | tzanger | Welcome to Cooking Up Some Slack. (CUSS) |
| 20:22.43 | tzanger | <PROTECTED> |
| 20:23.44 | *** join/#asterisk Gh0sty (~Ghosty@81.11.196.171) |
| 20:24.15 | Nugget | yay slack |
| 20:25.41 | mishehu | damn rsa auth |
| 20:26.27 | JerJer[interop] | mishehu: send the pastebin link again |
| 20:26.30 | JerJer[interop] | i missed it |
| 20:26.36 | JerJer[interop] | o0h found t |
| 20:26.38 | JerJer[interop] | it |
| 20:26.51 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 20:27.11 | ManxPower | I'm really starting to hate my new motherboard |
| 20:27.32 | JerJer[interop] | ok obvious question |
| 20:27.41 | JerJer[interop] | do you have the .key file loaded into asterisk? |
| 20:27.51 | JerJer[interop] | and the .pub file on the other box |
| 20:28.09 | ManxPower | JerJer[interop]: I told him to use secrets and have a simplier life. |
| 20:28.26 | JerJer[interop] | secrets have their limitations though |
| 20:28.40 | JerJer[interop] | soon we will move everyone to RSA |
| 20:28.59 | JerJer[interop] | to make things less stressfull on our end |
| 20:29.12 | ManxPower | JerJer[interop]: Why are secrets stressful? |
| 20:29.29 | JerJer[interop] | clear text for one |
| 20:29.46 | JerJer[interop] | you'd be surpized how many people have complained about that |
| 20:33.10 | JerJer[interop] | mishehu: i would remove trunk, at least for now |
| 20:33.10 | JerJer[interop] | keep it simple |
| 20:33.10 | ariel_ | JerJer[interop], I hate rsa keys |
| 20:33.10 | *** join/#asterisk nDuff (~cduffy@net-6621942-66.customer.corenap.com) |
| 20:33.10 | JerJer[interop] | ariel_: once you understand the implemenation in asterisk you will begin to like them |
| 20:33.11 | ariel_ | I will have to agree with ManxPower here |
| 20:33.11 | JerJer[interop] | when u have multple boxes, RSA auth is very nice |
| 20:33.11 | tzanger | rsa/dsa keys with ssh are just too cool for school |
| 20:33.12 | patdk | public key auth is the best |
| 20:33.12 | patdk | hmm, I wonder why that was underlined |
| 20:33.12 | Sedorox | it was? |
| 20:33.12 | JerJer[interop] | i just wish there was some way to deal with public key deployment and changes in asterisk |
| 20:33.12 | patdk | on my screen it is |
| 20:33.12 | ariel_ | JerJer[interop], my problem is not the key or how it's used. but how it's setup. |
| 20:33.55 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 20:34.15 | mgth | JerJer: you are smart, write an app :) |
| 20:34.36 | *** join/#asterisk delphi (~delphi@host81-152-229-127.range81-152.btcentralplus.com) |
| 20:34.38 | mishehu | gah |
| 20:34.42 | ariel_ | ManxPower, did you find your fall back person? |
| 20:34.53 | mishehu | JerJer[interop]: removing of trunk fixed it, now calls go both directions |
| 20:35.09 | JerJer[interop] | mishehu: because you did not have a valid type=peer on both ends |
| 20:35.32 | *** join/#asterisk thomas_adam (~n6tadam@host217-43-99-160.range217-43.btcentralplus.com) |
| 20:35.42 | mishehu | JerJer[interop]: actually, I do believe I do. I only posted a short snippet of the iax.conf |
| 20:35.59 | JerJer[interop] | if your logger.conf was setup sanely you would have gotten flodded with warnings saying that |
| 20:36.01 | delphi | hi, could anyone help we with a problem with zaptel not modprobing please? |
| 20:36.19 | JerJer[interop] | delphi: if you provide detail on your problem |
| 20:36.23 | JerJer[interop] | don't just ask to ask |
| 20:36.43 | patdk | heh, I wonder why there is so many people asking about zaptel installation problems lately |
| 20:36.51 | patdk | alot of new users, or just major changes |
| 20:37.20 | ariel_ | patdk, it comes in waves |
| 20:37.25 | delphi | JerJer[interop]: when modprobing i am getting a number of errors a long the lines of: /lib/modules/2.4.27-2-386/zaptel/zaptel.o: /lib/modules/2.4.27-2-386/zaptel/zaptel.o: unresolv |
| 20:37.25 | delphi | ed symbol devfs_unregister_R11457980 |
| 20:37.38 | JerJer[interop] | make clean install |
| 20:37.50 | nDuff | I've got an office w/ users accessing Asterisk via SIP phones. I'd like to provide them with a way to optionally mask their caller ID info when making outgoing calls (providing just the company's front-end number rather than their direct lines). I was pondering giving each phone (Sipura SPA-) a #2 extension with caller ID set differently (but voicemail and such still going to the same place) -- but I'd appreciate suggestions for other |
| 20:37.50 | nDuff | <PROTECTED> |
| 20:38.02 | JerJer[interop] | delphi: then make sure you have the approprate source for the runnng kernel |
| 20:38.15 | delphi | JerJer[interop]: ok, will try thanks |
| 20:38.35 | *** join/#asterisk verge (~jfargen@rrcs-24-227-48-10.se.biz.rr.com) |
| 20:38.40 | ariel_ | nDuff, just put the callerid in your outbound rules |
| 20:39.24 | delphi | JerJer[interop]: doen make clean install, and have the correct source, but it still does the same |
| 20:39.25 | ariel_ | nDuff, you can also set it up via different access numbers like 8 if they want theres or 9 if they want the co's number. |
| 20:40.00 | nDuff | ariel_, ahh -- that latter suggestion sounds 'bout right. |
| 20:40.16 | JerJer[interop] | delphi: when u get those messages it means you are linking against the wrong kernel |
| 20:40.28 | JerJer[interop] | kernel symbols are not correct |
| 20:41.17 | delphi | ok, i'll re-check |
| 20:42.18 | nDuff | ariel_, any hints wrt where I could look for docs relevant to setting that up? We don't use access numbers for dialing out presently, and were that implemented, I'm still not sure what mechanism to use to switch the caller ID based on it. |
| 20:42.59 | delphi | JerJer[interop]: everything does seem to match. i'm running debian sarge with a 2.4 stock kernel, any problems with that? |
| 20:43.12 | JerJer[interop] | look at the gcc command line - see if there is some funky directory getting included |
| 20:43.18 | JerJer[interop] | -I |
| 20:43.19 | JerJer[interop] | lines |
| 20:43.52 | ariel_ | nDuff, have you taken a look at the sample file. /usr/src/asterisk/configs/extensions.conf it has a pretty good way of doing it there with the 9 . |
| 20:44.02 | nDuff | ariel_, thanks, I'll look there. |
| 20:44.10 | ariel_ | sorry extensions.conf.sample |
| 20:45.19 | *** join/#asterisk riksta (~rick@81-178-193-191.dsl.pipex.com) |
| 20:46.22 | mishehu | http://pastebin.ca/8740 - shows what I believe to be the valid type=user and type=peer entries into the iax.conf's on either server. |
| 20:46.46 | mishehu | if somebody could look and verify if this is the case, I'd appreciate it. |
| 20:47.03 | delphi | JerJer[interop]: it's all fine. |
| 20:47.47 | JerJer[interop] | have you done at lease a make menuconfig on the kernel ? |
| 20:47.49 | JerJer[interop] | least |
| 20:49.00 | delphi | Jearil: no, make oldconfig |
| 20:49.17 | delphi | sorry, that should be JerJer[interop]: no, make oldconfig |
| 20:49.57 | JerJer[interop] | well whatever ...just configure your kernel |
| 20:50.08 | JerJer[interop] | i'm just guessng here |
| 20:50.47 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 20:51.00 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 20:52.04 | tzanger | make menuconfig please |
| 20:52.16 | tzanger | make oldconfig just brings the config file up to spec for the new kernel |
| 20:52.20 | tzanger | make menuconfig does a few other things too |
| 20:52.55 | L|NUX | getting this error while setting clone to x100p Apr 4 06:53:35 ERROR[1717]: chan_zap.c:6213 mkintf: Signalling requested is FXO Loopstart but line is in FXS Kewlstart signalling |
| 20:52.56 | L|NUX | Apr 4 06:53:35 ERROR[1717]: chan_zap.c:9148 setup_zap: Unable to register channel '1' |
| 20:52.56 | L|NUX | Apr 4 06:53:35 WARNING[1717]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 |
| 20:52.56 | L|NUX | <PROTECTED> |
| 20:52.56 | L|NUX | <PROTECTED> |
| 20:52.57 | L|NUX | Apr 4 06:53:35 WARNING[1717]: loader.c:440 load_modules: Loading module chan_zap.so failed! |
| 20:52.59 | L|NUX | Ouch ... error while writing audio data: : Broken pipe |
| 20:53.19 | mgth | L|nux: Please use a pastebin |
| 20:53.19 | tzanger | L|NUX: the error is very clear |
| 20:53.25 | L|NUX | sorry |
| 20:53.26 | tzanger | and pleae don't paste that here, use pastebin.ca |
| 20:53.28 | JerJer[interop] | read the first line |
| 20:53.32 | L|NUX | tzanger : ? |
| 20:53.47 | JerJer[interop] | that you flooded |
| 20:53.52 | mishehu | *sigh* |
| 20:54.18 | L|NUX | tzanger : sorry for that i will use pastebin.com can you please tell me what is the problem ? |
| 20:54.31 | ariel_ | L|NUX, you need to use fxs_ks instead of fxo_ks |
| 20:54.35 | mishehu | L|NUX: did you configure zaptel.conf? |
| 20:54.36 | JerJer[interop] | L|NUX: the very first line that you flooded is the problem |
| 20:54.43 | L|NUX | yea |
| 20:54.43 | mishehu | and zapata.conf |
| 20:54.56 | JerJer[interop] | an FXO device uses FXS signalling |
| 20:55.37 | L|NUX | hmmm |
| 20:55.38 | L|NUX | wait |
| 20:55.51 | L|NUX | fxsks=1 |
| 20:56.00 | L|NUX | have this in my /etc/zaptel.conf |
| 20:57.18 | JerJer[interop] | then u need fxs_ks in asterisk/zapata.conf |
| 20:57.28 | L|NUX | wait |
| 20:58.13 | L|NUX | k |
| 20:58.31 | L|NUX | work |
| 20:59.12 | *** join/#asterisk sigmounte (~sigmounte@lns-vlq-29-82-254-15-69.adsl.proxad.net) |
| 20:59.24 | JerJer[interop] | thank you, drive thru |
| 20:59.32 | marks__ | COVERTCALL IS FOR SALE.. http://covertcall.com/forums/viewtopic.php?t=278 |
| 20:59.57 | nine76 | wow |
| 21:00.31 | *** join/#asterisk Slainte (~Slainte@66.55.112.85.ppp.northrock.bm) |
| 21:00.35 | L|NUX | but i have problem |
| 21:01.07 | nine76 | If anyone has a second please look at this and give me any input on fixing it:-/ http://pastebin.ca/8742 |
| 21:01.27 | L|NUX | then i can't dial any number :( |
| 21:01.28 | tzanger | nine76: how about you give us a 30-word explanation of what it is so we can decide whether to bother clicking or not |
| 21:01.33 | nine76 | k |
| 21:01.35 | *** join/#asterisk Carp1 (carp_xigon@204.97.151.254) |
| 21:01.36 | nine76 | fair enough |
| 21:02.11 | JerJer[interop] | L|NUX: make an extension to dial then |
| 21:02.29 | L|NUX | wait |
| 21:02.39 | L|NUX | exten => _X.,1,Dial(Zap/1/${EXTEN}) |
| 21:02.41 | L|NUX | i added this |
| 21:02.47 | L|NUX | wait |
| 21:02.49 | L|NUX | let me try again |
| 21:03.22 | nine76 | I got areskicc installed and going. Many many error fixes later, it now TX's and RX's. But when called it exits 0. http://pastebin.ca/8742 is the console output with agi debug on. I think to someone who uses areskicc,they may be familiar with what the next step would be. |
| 21:03.52 | tzanger | nine76: that is EXACTLY how people should ask for help here. That is a perfect example |
| 21:03.56 | tzanger | unfortunately I don't use asteriskcc |
| 21:04.01 | niZon | does anyone know of a provider simmilar to iax.cc? |
| 21:04.19 | niZon | preferrably one that offers 204 DIDs |
| 21:04.21 | nine76 | I asked that way last night and couldnt find help. I figrued I would try this a.m. and then take it to the lists. |
| 21:05.12 | nDuff | Can I configure *67 to provide the company's front-office phone# rather than disabling caller ID altogether? |
| 21:05.19 | niZon | AGI is evil |
| 21:05.24 | shido6 | u can do whatever u vant |
| 21:05.36 | shido6 | vut ever you vant |
| 21:05.37 | JerJer[interop] | AGI is very evil |
| 21:05.59 | JerJer[interop] | nine76: write your own calling card app - its so simple it is almost trivial |
| 21:06.00 | nine76 | areski stat v2 installed very painlessly,looks good. id recommen it to everyone. |
| 21:06.15 | Carp1 | link please. |
| 21:06.33 | JerJer[interop] | the tough part is billing, whch areskicc does not deal with so you are better off writing your own |
| 21:06.36 | *** join/#asterisk kraeMit (~chatzilla@p54892FC5.dip0.t-ipconnect.de) |
| 21:06.39 | nine76 | I started too JerJer. 10 hrs later I decide to try whats already available. even switched my db's to postgres... |
| 21:06.47 | nine76 | one second on link Carpl |
| 21:06.48 | tzanger | nDuff: of course you can; you can make * do damn near anything you want |
| 21:06.52 | JerJer[interop] | what is available is crap |
| 21:06.53 | tzanger | it's all a matter of the dialplan logic |
| 21:06.56 | Carp1 | ok |
| 21:07.11 | Carp1 | nine76: please PM it to me, I have to run for like 10 minutes. |
| 21:07.12 | nine76 | http://areski.net/asterisk-stat-v2/ |
| 21:07.14 | Darwin[laptop] | hell * will even scrw your dog if you set it up right |
| 21:07.15 | Carp1 | nevermind |
| 21:07.18 | Carp1 | I got it |
| 21:07.19 | Carp1 | thanks. |
| 21:07.25 | nine76 | I'm not *that* slow:) |
| 21:07.47 | *** join/#asterisk fugitivo (~ajf@201.255.100.195) |
| 21:07.51 | fugitivo | hello |
| 21:08.20 | *** join/#asterisk omelia (~jana_009@pc-66-208-83-200.cm.vtr.net) |
| 21:08.35 | omelia | hola |
| 21:08.55 | omelia | holaa |
| 21:09.15 | nine76 | Would it not be better to use areskicc as a base then build on additional features? |
| 21:09.49 | JerJer[interop] | nine76: no |
| 21:09.53 | JerJer[interop] | AGI does not scale |
| 21:10.03 | omelia | q idioma ablan? |
| 21:10.04 | nine76 | i see |
| 21:10.16 | omelia | ?? |
| 21:10.23 | tzanger | nDuff: personally I'd have *67 setcidnum and then read() and goto the normal context with exten of whatever was read |
| 21:10.24 | moy | aqui ingles la mayoria |
| 21:10.30 | omelia | aaa |
| 21:10.34 | tzanger | nDuff: that's just an "off the top of my head" solution |
| 21:10.45 | moy | si quieres español puedes usar el canal php-es |
| 21:10.49 | cjk | hi, do you guys know any mean to detect in someway where the user is registering from and base on this information redirect the registration stuff to a server in his country..... i do not need the software or the solution, just the concept |
| 21:10.51 | omelia | what are you talking about?? |
| 21:11.11 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 21:11.16 | omelia | aa gx |
| 21:11.20 | cjk | at the moment i only see DNS as a possible solution |
| 21:12.31 | Slainte | cjk, do a lookup on what country their IP is registered to, and that should work most of the time. |
| 21:12.46 | JerJer[interop] | cjk: i cannot see why would anyone care |
| 21:13.20 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 21:13.23 | JerJer[interop] | Slainte: that information is not reliable |
| 21:13.31 | JerJer[interop] | espically with portable IP blocks |
| 21:13.36 | Slainte | more reliable then dns |
| 21:13.44 | *** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 21:13.55 | brycec | Any asterisk/zaptel devs on here? |
| 21:13.55 | JerJer[interop] | just design a system that doesn't care what country they are in |
| 21:14.10 | L|NUX | i have problem |
| 21:14.13 | nine76 | lol |
| 21:14.21 | marlowe | by: Just ask your question |
| 21:14.25 | JerJer[interop] | brycec: no this is a forum for the disucssion of the ascii charector called * |
| 21:14.26 | marlowe | L: Just ask your question |
| 21:14.26 | nine76 | What would that be L|NUX |
| 21:14.29 | L|NUX | when i try to call to my local number operator says the number you dialed is invalied :) |
| 21:14.31 | delphi | tzanger: compiled a new kernel, but still getting the same error messages |
| 21:14.39 | L|NUX | s/invalid |
| 21:14.44 | brycec | haha, thanks JerJer[interop] |
| 21:14.50 | nine76 | look in console to see exactly what number was dialed |
| 21:14.54 | JerJer[interop] | L|NUX: then call a valid number |
| 21:14.56 | *** part/#asterisk cia (~cwj@adsl-68-77-11-148.dsl.emhril.ameritech.net) |
| 21:15.00 | brycec | JerJer[interop], the topic would indicate otherwise :-P |
| 21:15.04 | L|NUX | well i am calling valid number |
| 21:15.12 | JerJer[interop] | not accordng to the operator |
| 21:15.13 | L|NUX | do i need to dial some thing like 91 then call ? |
| 21:15.14 | nine76 | check console to make sure its attempting to call a valid # |
| 21:15.17 | Slainte | "portable IP blocks?" They need to be associated with a BGP AS number, and you can see what ISP is gatewaying that IP block |
| 21:15.19 | *** join/#asterisk [1]jakepdev (~jakepdev@pool-70-16-137-171.phil.east.verizon.net) |
| 21:15.27 | cjk | Slainte, doing the lookup is easy |
| 21:15.29 | cjk | and then |
| 21:15.32 | cjk | what do i do then |
| 21:15.36 | JerJer[interop] | Slainte: and how quickly is that going to be determined? |
| 21:15.39 | JerJer[interop] | then what if it changes? |
| 21:15.44 | cjk | whats the technique to tell my phone to register as ip x.x.x.x |
| 21:15.44 | JerJer[interop] | your just asking for trouble |
| 21:15.58 | brycec | I'm having issues with DTMF recognition, as in there is none apparently being done, on a zaptel device. ztmonitor shows the audio is received at least |
| 21:16.03 | JerJer[interop] | cjk: register with proxy |
| 21:16.20 | ariel_ | strange network problems for my connection today! |
| 21:16.22 | JerJer[interop] | brycec: what kind of DTMF indication? |
| 21:16.32 | cjk | JerJer[interop], ok so i need to register at an SER which has no idea about what iax is |
| 21:16.46 | JerJer[interop] | ok and this is a problem how? |
| 21:16.51 | JerJer[interop] | asterisk talks sip |
| 21:17.03 | JerJer[interop] | and one can register to asterisk with sip |
| 21:17.04 | brycec | JerJer[interop], Nothing on the channel's debug and no break of dialtone |
| 21:17.07 | cjk | i know, but maybe i want to talk iax |
| 21:17.12 | JerJer[interop] | then talk IAX |
| 21:17.13 | L|NUX | any idea |
| 21:17.42 | JerJer[interop] | iax has the same registration concept |
| 21:18.00 | L|NUX | i added exten => _X.,1,Dial(ZAP/1/${EXTEN}) |
| 21:18.06 | L|NUX | and restared asterisk |
| 21:18.21 | JerJer[interop] | that's evil |
| 21:18.25 | L|NUX | and when i try to dial any local number operator say no invalid |
| 21:18.25 | L|NUX | :( |
| 21:18.29 | L|NUX | any hope ? |
| 21:18.32 | *** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:18.40 | cjk | JerJer[interop], im not sure if i understant the job of a proxy, but the traffic always goes through a proxy |
| 21:18.46 | moy | what does the console says LINUX? |
| 21:18.51 | L|NUX | wait |
| 21:18.53 | nine76 | look at your console to see what number its attempting to dial L|NUX |
| 21:19.11 | nine76 | You should be able to then obviously see the problem,and correct it in exten.conf |
| 21:19.17 | moy | agree |
| 21:19.20 | JerJer[interop] | _1NXXNXXXXXX is more proper for a us/canada number |
| 21:20.09 | tzanger | do not use . unless you ABSOLUTELY must... |
| 21:20.28 | L|NUX | hmm |
| 21:20.36 | L|NUX | <PROTECTED> |
| 21:20.36 | L|NUX | <PROTECTED> |
| 21:20.36 | L|NUX | <PROTECTED> |
| 21:20.38 | L|NUX | got this |
| 21:20.52 | nine76 | it only tried to dial 6 number? |
| 21:20.57 | nine76 | read it :|NUX |
| 21:21.10 | nine76 | 789744? |
| 21:21.21 | nine76 | missing a digit,check for EXTEN:1 in exten.conf |
| 21:21.43 | *** join/#asterisk StallmanIsGod (~mindCrime@rrcs-24-106-188-6.se.biz.rr.com) |
| 21:21.50 | L|NUX | wait |
| 21:21.53 | nine76 | lol |
| 21:21.59 | JerJer[interop] | or user didn't dial enough digitis |
| 21:22.15 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
| 21:22.21 | L|NUX | i have this exten => _X.,1,Dial(ZAP/1/${EXTEN:1}) |
| 21:22.25 | nine76 | remove :1 |
| 21:22.29 | Carp1 | nine |
| 21:22.30 | L|NUX | k |
| 21:22.32 | nine76 | so ${EXTEN} |
| 21:22.43 | nine76 | try it that way,and watch and READ console |
| 21:22.48 | brycec | sob0l, can nobody help me with zaptel/asterisk dtmf recognition??? |
| 21:22.52 | nine76 | Hi Carpl |
| 21:23.02 | Carp1 | Hey. |
| 21:23.09 | Carp1 | nevrmind lol |
| 21:23.10 | L|NUX | hmm |
| 21:23.11 | L|NUX | works |
| 21:23.11 | L|NUX | ;) |
| 21:23.12 | Carp1 | Thought I lost hte link |
| 21:23.15 | brycec | So can nobody help me with zaptel/asterisk dtmf recognition??? |
| 21:23.21 | Carp1 | tru |
| 21:23.23 | L|NUX | nine76 : thx |
| 21:23.28 | *** join/#asterisk JerJer[mobile] (1000@dhcp-11-147.hstf.interop.net) |
| 21:23.43 | nine76 | brycec: x100p? |
| 21:23.46 | L|NUX | nine76 : how can i bound some one that only that extension can make outbound calls ? |
| 21:24.02 | nine76 | I would do it using contexts |
| 21:24.05 | brycec | nine76, yeah |
| 21:24.24 | L|NUX | nine76 : can you tell me how ? |
| 21:24.25 | brycec | nine76, er, rather te100 |
| 21:24.56 | brycec | It works on x86, but not ppc |
| 21:25.09 | nine76 | brycec: I have an x100p and never experienced dtmf problems,so I was just going to recommend looking over config files again,but if your not using the same card as I, I do not know:-/ |
| 21:25.24 | brycec | nine76, You on ppc too? |
| 21:25.30 | nine76 | no |
| 21:25.44 | brycec | nine76, Yeah, it works on x86 just fine for me |
| 21:26.14 | L|NUX | nine76 : can you tell me how can i restrict only one or two extenstion to make outbound calls ? |
| 21:26.35 | nine76 | I told you I would use different contexts for that purpose. |
| 21:26.44 | L|NUX | like |
| 21:26.51 | L|NUX | i am new man :$ |
| 21:26.54 | brycec | L|NUX, Simply put them in a context that has no includes with contexts that permit outbound lines |
| 21:27.03 | nine76 | ^^^ |
| 21:27.03 | blitzrage | contexts are security boundries. Use them, understand them, love them. |
| 21:27.33 | L|NUX | can i show you my configurations ? |
| 21:27.45 | L|NUX | :) |
| 21:27.52 | _SMP_ | anyone in here own a 7960? How loud is your handset volume if you crank it all the way up? In order to get decent vol, I have to crank it almost to the limit. Is that normal? |
| 21:28.23 | *** join/#asterisk netMonkey (~netMonkey@209.8.233.249) |
| 21:28.49 | nine76 | Its no error on your part L|NUX no need to look at configs. You need to understand what contexts are,since its what you need. |
| 21:28.52 | Nugget | it's uncomfortably loud if I crank it all the way up |
| 21:28.54 | blitzrage | create contexts related to various features ([voicemail], [outbound-pstn], [inbound-nufone], etc...). Then create contexts [basic], [trusted] and [administrator]. Add include => context to each of those, which gives you levels of control. Then assign your phones to either context=basic|trusted|administrator |
| 21:29.25 | L|NUX | ok |
| 21:29.35 | blitzrage | * context crash course brought to you by blitzrage * :) |
| 21:29.46 | nine76 | well done |
| 21:29.50 | nine76 | :) |
| 21:29.53 | blitzrage | lol |
| 21:29.59 | L|NUX | :> |
| 21:30.24 | blitzrage | say it outloud: contexts are security boundaries |
| 21:31.03 | file[laptop] | contexts are security boundaries! |
| 21:31.07 | JerJer[interop] | _SMP_: ditto - i've got mine less than half way up |
| 21:31.08 | blitzrage | file[laptop]: @ |
| 21:31.11 | blitzrage | errr... ! |
| 21:31.23 | file[laptop] | lol |
| 21:31.57 | tzanger | contexts kick ass |
| 21:32.10 | fugitivo | anyone using wireless headsets? |
| 21:32.27 | blitzrage | tzanger: I prefer everything in default |
| 21:32.37 | nine76 | lol |
| 21:32.47 | blitzrage | tzanger: especially int'l calling through my PRI |
| 21:33.11 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 21:33.12 | blitzrage | JerJer[interop]: can I have one? :) |
| 21:33.19 | nine76 | beat em to it |
| 21:33.26 | JerJer[interop] | sure, if your in Michigan |
| 21:33.43 | blitzrage | JerJer[interop]: actually, you can just colocate it for me, I don't need physical access :) |
| 21:33.49 | blitzrage | thanks! |
| 21:34.30 | *** join/#asterisk file[laptop] (~file@mctn1-3451.nb.aliant.net) |
| 21:35.55 | JerJer[interop] | sure send a box |
| 21:37.10 | JerJer[interop] | and if it has two NICs you won't need to put a T1 card in it |
| 21:37.28 | JerJer[interop] | since we have a private IP network for voice only - where usage of ulaw is encuraged |
| 21:37.42 | shido6 | damn right |
| 21:37.46 | shido6 | das blinke lights |
| 21:37.50 | shido6 | das blinken lights |
| 21:37.53 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 21:37.55 | fearnor | well |
| 21:37.58 | fearnor | TDMoE isn't really TDM :) |
| 21:37.59 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 21:38.01 | JerJer[interop] | more blinken |
| 21:38.04 | fearnor | close, but no cigarre |
| 21:38.18 | JerJer[interop] | there is no TDMoE |
| 21:38.22 | JerJer[interop] | just IAX2 |
| 21:38.29 | blitzrage | JerJer[interop]: lol, are you serious? |
| 21:38.29 | fearnor | well, then no fax for joo |
| 21:38.35 | fearnor | that's gay. sorry. |
| 21:38.51 | mishehu | relaxen und watchin das blinken lights |
| 21:38.52 | fearnor | half my clients buy PRIs from me just so they can do faxing :) |
| 21:39.13 | fearnor | mishehu: das buttonen ist nicht fur gefingerpoken und mittengrabben |
| 21:39.28 | fearnor | i can probably do that |
| 21:39.31 | mishehu | ich bin du |
| 21:39.38 | fearnor | have cisco 4000 collecting dust |
| 21:39.44 | fearnor | with 3*8 port bri :) |
| 21:39.51 | mishehu | fearnor: I don't speak that language |
| 21:39.52 | mishehu | heh |
| 21:40.00 | *** join/#asterisk Helloween22 (~xxxx@200.115.203.91) |
| 21:40.03 | *** part/#asterisk Helloween22 (~xxxx@200.115.203.91) |
| 21:40.06 | blitzrage | du haste miche! |
| 21:40.09 | L|NUX | blitzrage : now see what i have done :) |
| 21:40.16 | fearnor | du hasst |
| 21:40.17 | L|NUX | contexts |
| 21:40.24 | L|NUX | [admin] |
| 21:40.32 | L|NUX | wait i use pastebin.com |
| 21:40.38 | shido6 | Sie mögen nicht meine Blinkenlichter? |
| 21:40.40 | blitzrage | L|NUX: better idea ;) |
| 21:40.58 | mishehu | ðå... |
| 21:41.35 | JerJer[interop] | du hast |
| 21:41.45 | fearnor | jerjer: hast is have, hasst is hate. |
| 21:41.52 | fearnor | i think. |
| 21:41.59 | JerJer[interop] | just sounded good :) |
| 21:42.07 | fearnor | definitely do hasst |
| 21:42.12 | fearnor | :) |
| 21:42.13 | blitzrage | fearnor: yes, hate :) |
| 21:42.21 | fearnor | hate flows freely |
| 21:42.27 | L|NUX | blitzrage: http://www.pastebin.com/266660 |
| 21:42.32 | shido6 | Ich mag gerade den deutschen Schokoladenkuchen |
| 21:42.55 | L|NUX | but when i dialing from extension 1001 it said the person you are calling is unavilable |
| 21:43.31 | cjk | just to come back to my problems, what to you think of "sip redirect servers" which give a different answer based on ip information |
| 21:44.40 | Dovid | hello all |
| 21:44.59 | Dovid | i set up a basic asterisk machine and i need a basic softphone. can anyhone assist ? |
| 21:45.09 | nine76 | kphone,x-lite |
| 21:45.13 | nine76 | many available |
| 21:45.45 | Dovid | i need one for windows |
| 21:45.49 | nine76 | x-lite |
| 21:45.53 | L|NUX | blitzrage : any idea ? |
| 21:45.58 | cjk | Dovid, sjlabs.com a really good one |
| 21:46.31 | fugitivo | cjk: the interface of sjphone is just ugly for linux or i'm missing something? |
| 21:46.36 | L|NUX | xten.com are best :) |
| 21:46.42 | cjk | fugitivo, its different on windows |
| 21:46.45 | cjk | big difference |
| 21:46.50 | cjk | really really big difference |
| 21:46.57 | cjk | forget the linux version of sjlabs |
| 21:48.27 | JerJer[interop] | sweeet - they are doing bluetooth to a Zultys 4x5 SIP phone here |
| 21:48.34 | nine76 | x-lite windows version works on linux using crossover office |
| 21:48.46 | nine76 | meanwhile linux x-lite doesnt work for me:-/ |
| 21:50.11 | cjk | x-lite sucks, sorry i think its really unstable and is missing options in the free version |
| 21:50.18 | cjk | sjlabs has all the important options |
| 21:50.19 | *** join/#asterisk bizbaz (bizbaz@66-215-223-162.riv-eres.charterpipeline.net) |
| 21:50.26 | bizbaz | hi |
| 21:50.37 | bizbaz | can any one help me? |
| 21:50.40 | ManxPower | All softphones suck! |
| 21:50.50 | fugitivo | ManxPower: i think you're right, he |
| 21:50.56 | nine76 | I dislike x-lite myself,cant find better alternative though. kphone lieks to seg fault at random. I dont use windows. |
| 21:51.04 | bizbaz | who knows any thing about unlockig phones |
| 21:51.13 | ManxPower | bizbaz: We are not going to configure Asterisk for you, but if you have a specific question someone may be able to help. |
| 21:51.16 | fugitivo | kphone is nice, but yet too buggy |
| 21:51.28 | bizbaz | oo ok |
| 21:51.33 | delphi | kiax seems to work quite well |
| 21:51.34 | ManxPower | fugitivo: kphone doesn't support OOB DTMF |
| 21:52.14 | ManxPower | nine76: Make your live better - buy a hardphone |
| 21:52.17 | fugitivo | ManxPower: i know, but it's the nicest for linux |
| 21:52.20 | nine76 | agreed! |
| 21:52.34 | nine76 | Hope to purchase a sipura soon. |
| 21:52.37 | bizbaz | i just want to know if it is possible to use a nextl i860 with an at&t sim car |
| 21:53.02 | fugitivo | did you see the lastest version of cisco softphone? |
| 21:53.04 | fugitivo | with video? |
| 21:53.16 | fugitivo | that's a nice softphone :) |
| 21:53.17 | Sedorox | bizbaz: no |
| 21:53.18 | bizbaz | do i need to unlock it? |
| 21:53.27 | Sedorox | no |
| 21:53.28 | Sedorox | different systems |
| 21:53.33 | ManxPower | bizbaz: that's not an Asterisk question, but the answer is No. Nextel phones don't work with any other carrier |
| 21:53.43 | Sedorox | I tried using cingular on a i730... didn't work.. |
| 21:53.45 | bizbaz | oo ok |
| 21:53.51 | Sedorox | conected.. but couldn't make calls |
| 21:54.12 | ManxPower | Sedorox: As long as the phone and the provider use the same tech it CAN work, but carriers don't like to do that. |
| 21:54.38 | bizbaz | thats right but what is you unlock it |
| 21:54.57 | ManxPower | bizbaz: As I said Nextel uses their own protocol. |
| 21:54.58 | bizbaz | then its just the phone |
| 21:54.59 | Sedorox | well I couldn't get it to work.. so just talking from experience.. and yes.. the phone I tried is unlocked... |
| 21:55.17 | bizbaz | i see |
| 21:55.36 | L|NUX | can some one please tell me why my extenstion 1001 can't make outbound calls |
| 21:55.46 | ManxPower | Europe is so much nice for this stuff. All phones use the same protocols and carriers do allow you to keep your phone. |
| 21:56.02 | L|NUX | i have this configuration in my extensions.conf http://www.pastebin.com/266660 |
| 21:56.04 | nine76 | L|NUX what context is your sip phone placed into? |
| 21:56.15 | bizbaz | i wish i could use this phone |
| 21:56.19 | ManxPower | L|NUX: You are saing the equiv of "Can someone please tell me why my car doesn't work." I.e. it's too general of a question |
| 21:56.36 | L|NUX | [admin] |
| 21:56.42 | ManxPower | Hell, as far as I know Nextel phones don't even HAVE a SUM card. |
| 21:56.46 | BuckRogers | did you check your oil? |
| 21:56.49 | BuckRogers | j.k. |
| 21:56.54 | Sedorox | nextel's do run sims |
| 21:57.02 | ManxPower | SUM == SIM |
| 21:57.03 | L|NUX | as well as in [sip] |
| 21:57.05 | fugitivo | ManxPower: yes they have sim cards |
| 21:57.06 | BuckRogers | i hate my nextel cant wait for the contract to be up |
| 21:57.14 | bizbaz | yes they do they just use iden system |
| 21:57.22 | Sedorox | thats one thing that is gonna piss me off when sprint takes over.. their a non-sim system |
| 21:57.38 | fugitivo | sprint is too expensive |
| 21:57.39 | nine76 | L|NUX in sip.conf your 1001 has a line which says context => admin ? |
| 21:57.43 | ManxPower | Sedorox: Honestly, IDEN must die. |
| 21:57.53 | L|NUX | nah |
| 21:57.58 | L|NUX | do i need to add this ? |
| 21:58.00 | L|NUX | wait |
| 21:58.01 | L|NUX | let me add |
| 21:58.02 | bizbaz | thats way im trying to get my at&t sim to work on the nextl |
| 21:58.19 | ManxPower | bizbaz: ask on a cell phone channel |
| 21:58.33 | Sedorox | I like iden... |
| 21:58.57 | ManxPower | Sedorox: We already have too many cell phone standards in the USA. |
| 21:58.58 | bizbaz | do you know of any one? |
| 21:59.00 | cjk | hi, anyone here who can tell me if a redirect server is able to redirect registration requests as well. i could not really get that our of the rfc and google |
| 21:59.05 | L|NUX | not working :( |
| 21:59.10 | ManxPower | iDEN, GSM, CDMA, TDMA, AMPS |
| 21:59.10 | Sedorox | Anyway... bizbaz if you decide to sell your phone.. let m,e know :-p |
| 21:59.15 | Sedorox | ManxPower: true |
| 21:59.21 | nine76 | get console output and put it on pastebin and gimme link L|NUX |
| 21:59.31 | L|NUX | nothing here |
| 21:59.35 | ManxPower | Fortunatly all carriers are switching to CDMA or GSM |
| 21:59.36 | L|NUX | Asterisk Ready. |
| 21:59.44 | bizbaz | ill give it to you for 150 |
| 21:59.53 | ManxPower | still not a single standard, but better than 5 of them |
| 22:00.08 | Sedorox | yea.. but I like the idea of using SIM's because I don't have to go back to the carrier to switch phones |
| 22:00.35 | ManxPower | Sedorox: The ONLY reason I don't have a GSM phone right now is that 2 years ago the GSM carrier's coverage SUCKED. |
| 22:00.42 | ManxPower | I believe in the technology |
| 22:00.46 | Sedorox | ah |
| 22:01.21 | Sedorox | I just wish CDMA would do SIM cards... |
| 22:01.24 | Sedorox | makes it easier on me |
| 22:02.07 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 22:02.51 | ManxPower | I was looking at cell phones on eBay (for Europe) and the prices are like 5x what they are for phones in the USA |
| 22:03.01 | shmaltz | just received my first quad span t1 from digium, anybody here know what the jumpers are for? |
| 22:03.27 | Sedorox | that sucks.. can't you just use a gsm/world phone from the US? |
| 22:03.28 | ManxPower | shmaltz: forcing T-1/E-1. |
| 22:03.42 | ManxPower | Sedorox: Well, if I HAD a GSM/World phone.... |
| 22:03.53 | ManxPower | Sedorox: I was referring to the prices of USED phones. |
| 22:04.00 | Sedorox | oo ok |
| 22:04.07 | shmaltz | ManxPower, what about the timing jumpers? |
| 22:04.23 | ManxPower | shmaltz: no idea. don't touch them |
| 22:04.34 | shmaltz | I'm not just asking |
| 22:04.39 | shmaltz | and the test jumpers? |
| 22:04.52 | shmaltz | and whats the expansion slots for? |
| 22:05.05 | ManxPower | shmaltz: you sure do ask a lot of questions |
| 22:05.15 | shmaltz | hey it's sunday |
| 22:05.18 | shmaltz | ;) |
| 22:05.53 | fugitivo | shmaltz: the manual doesn't give you that info? |
| 22:05.55 | kraeMit | To be honest: Here is is monday since 5 minutes ;-) |
| 22:06.05 | ManxPower | fugitivo: there is no manual |
| 22:06.21 | shmaltz | kraeMit, i'm glad I live in the west |
| 22:06.22 | fugitivo | ManxPower: no? why? |
| 22:06.47 | kraeMit | ;-) |
| 22:06.55 | ManxPower | fugitivo: I don't know. PRinting one would add like $1 to the cost of the product |
| 22:09.43 | *** join/#asterisk JerJer[mobile] (1000@dhcp-11-147.hstf.interop.net) |
| 22:09.45 | shmaltz | wow, gmail expanded storage to 2 GB |
| 22:09.56 | fugitivo | shmaltz: old news :) |
| 22:10.08 | *** join/#asterisk deRost (~deRost@054.209-89-66-0.interbaun.com) |
| 22:10.10 | tzanger | shmaltz: are you anywhere near your 1g now?? |
| 22:10.10 | shmaltz | well I just noticed |
| 22:10.14 | file[laptop] | they keep going up |
| 22:10.16 | shmaltz | nah |
| 22:10.16 | file[laptop] | at 2055 now |
| 22:10.19 | hmodes | hrmm |
| 22:10.20 | file[laptop] | 2055MB |
| 22:10.41 | fugitivo | tzanger: You are currently using 748 MB (36%) of your 2055 MB. |
| 22:10.42 | shmaltz | gmail reports: |
| 22:10.43 | hmodes | is there a way to listen for dtmf during a call and perform actions? |
| 22:10.44 | shmaltz | *You are currently using 134 MB (7%) of your 2055 MB.* |
| 22:10.46 | Slainte | anyone using the sql billing option? |
| 22:10.53 | hmodes | it looks like a macro should do what I want, but doesn't seem to |
| 22:10.55 | fugitivo | hmodes: yes |
| 22:11.00 | tzanger | damn |
| 22:11.24 | hmodes | fugitivo: care to drop a hint as to what I should be searching for? :) |
| 22:12.39 | shmaltz | anybody here that qualifies, and is interested in designing a gui to manage: |
| 22:12.41 | shmaltz | extensions.conf, sip.conf, voicemail.conf, musiconhold.conf, and is context aware? |
| 22:12.42 | shmaltz | will pay. |
| 22:15.11 | fugitivo | hmodes: http://www.voip-info.org/wiki-Asterisk+cmd+Read |
| 22:15.12 | ManxPower | shmaltz: HAHAHAHAHA!!!!!!!!!!!!! |
| 22:15.28 | fugitivo | shmaltz: why? vi is nice |
| 22:15.31 | shmaltz | ManxPower, why you laughing? |
| 22:15.32 | ManxPower | shmaltz: if it was easy someone would have done it already. |
| 22:15.46 | ManxPower | Hell, even if it was medium difficult someone would have done it already. |
| 22:15.47 | *** join/#asterisk Dovid (~hirisk@pool-151-198-12-130.mad.east.verizon.net) |
| 22:15.50 | shmaltz | I'm trying to give this to a client |
| 22:16.20 | shmaltz | I love vi, but my client doesn't know what it stands for |
| 22:16.21 | ManxPower | shmaltz: I would consider it for US$25,000 since it would take about 6 months worth of work to write and debug it. |
| 22:16.39 | shmaltz | ManxPower, I don't think it will take that long |
| 22:16.45 | shmaltz | I think 2 weeks is enough |
| 22:16.55 | shmaltz | pm me and I'll explain |
| 22:16.55 | Blissex | shmaltz: then go ahead :-) |
| 22:17.00 | ManxPower | shmaltz: The project is MUCH mroe complicated than you think it is. |
| 22:17.16 | file[laptop] | isn't that just cute |
| 22:17.18 | fearnor | haha |
| 22:17.20 | fugitivo | shmaltz: see if you can find something here, done it already. |
| 22:17.20 | fugitivo | <ManxPower> Hell, even if it was medium difficult someone would have done it already. |
| 22:17.20 | fugitivo | -:- Dovid [~hirisk@pool-151-198-12-130.mad.east.verizon.net] has joined #asterisk |
| 22:17.20 | fugitivo | <shmaltz> I'm trying to give this to a client |
| 22:17.22 | shmaltz | ManxPower, not for what I need |
| 22:17.24 | fugitivo | ouch |
| 22:17.25 | fugitivo | sorry |
| 22:17.28 | fearnor | start on it, shmaltz |
| 22:17.32 | fugitivo | http://freshmeat.net/search/?q=asterisk§ion=projects&Go.x=0&Go.y=0 |
| 22:17.34 | fugitivo | there |
| 22:17.38 | fearnor | and then you'll realize its far more complicated than you think |
| 22:17.40 | ManxPower | My clients call me for changes |
| 22:17.45 | fearnor | one of those 80/20 things |
| 22:17.52 | fearnor | 80% of work take 80% of time |
| 22:17.59 | fearnor | the other 20% of work take the other 80% of time. |
| 22:17.59 | fearnor | :) |
| 22:18.36 | *** join/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
| 22:19.04 | fearnor | good to know |
| 22:20.00 | ManxPower | I don't suppose anyone knows how much the small electric burners cost for a stove? |
| 22:21.55 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:21.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:23.10 | *** join/#asterisk cagundena (~chatzilla@7.Red-80-32-26.pooles.rima-tde.net) |
| 22:23.29 | ManxPower | I didn't think so |
| 22:23.54 | cagundena | hi |
| 22:25.03 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 22:26.15 | *** part/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
| 22:27.05 | robl^ | ManxPower: replacements? if its standard coil type (not the solid ones), then about $20-30 |
| 22:28.14 | *** join/#asterisk shodan (~shodan@216.113.99.219) |
| 22:29.42 | ManxPower | robl^: thanks. |
| 22:30.12 | *** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.hsd1.ca.comcast.net) |
| 22:30.23 | ManxPower | Note to self: check for plastic bags sitting on the top of the stove when you turn on the oven, expecially plastic bags sitting on top of the burner with the oven vent. |
| 22:30.25 | robl^ | ManxPower: should I ask how you broke one? |
| 22:30.35 | robl^ | ewww |
| 22:30.45 | fearnor | mmm smell of burnt plastic |
| 22:30.54 | ManxPower | robl^: Well, it involved rope and j-lube. |
| 22:30.55 | fearnor | teh magic smoke |
| 22:30.59 | robl^ | actually.. that can be cleaned off easily.. should still be ok |
| 22:31.02 | ManxPower | fearnor: just melted, not burnt. |
| 22:31.15 | ManxPower | For $20 I'll try to clean it off myself. |
| 22:31.28 | ubergoober | Has anybody had trouble using IAX2 with FWD today? |
| 22:31.36 | fearnor | just 20$? it ain't white man's job to clean it |
| 22:31.51 | fearnor | hire a mexican housekeeper let her deal wif it |
| 22:31.56 | *** join/#asterisk imediax (imediax@00045a809589.click-network.com) |
| 22:32.00 | robl^ | hehe.. just use some steel wool and a slave :) |
| 22:32.02 | fearnor | or buy new |
| 22:32.17 | fearnor | steel wool + computers = fun |
| 22:33.10 | *** join/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 22:33.34 | *** part/#asterisk neopher (~crazy@mail.techhelpresources.com) |
| 22:34.01 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.res.rr.com) |
| 22:35.47 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
| 22:36.39 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-197-92.dsl.scarlet.be) |
| 22:40.36 | *** join/#asterisk JerJer[mobile] (1000@dhcp-11-147.hstf.interop.net) |
| 22:43.38 | hmodes | hrmm, okay |
| 22:43.42 | hmodes | i am not getting this |
| 22:44.19 | JerJer[interop] | hmodes: nobody is getting it |
| 22:44.25 | hmodes | heh |
| 22:45.13 | marks__ | http://img136.exs.cx/my.php?loc=img136&image=pots7mc.jpg |
| 22:45.26 | *** join/#asterisk Godsey (lanny@goofball.md5.com) |
| 22:46.54 | hmodes | so i want an incoming call to dial a phone, and while the call is up, if 1 is pressed, execute a system action |
| 22:47.01 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 22:47.02 | hmodes | https://matrix.gs/door.txt |
| 22:47.09 | hmodes | that does not work, both with waitexten and read |
| 22:47.17 | MrBelvedr | did my last question go through? i got disconnected. |
| 22:47.23 | hmodes | the calling party just gets endless ringing, and when the dialed party picks up there is no audio |
| 22:47.40 | Carp1 | Anyone know the SIP server for nuFone? |
| 22:47.41 | hmodes | anyone have any suggestions? |
| 22:47.56 | MrBelvedr | what is the minimum RAM that asterisk can run under? assuming no transcodding and no Music on hold or anything) |
| 22:47.57 | shido6 | yeah |
| 22:48.00 | shido6 | whats up Carp1 |
| 22:48.01 | shido6 | ? |
| 22:48.07 | Carp1 | Not much, you? |
| 22:48.11 | Carp1 | Just need the SIP server. |
| 22:48.53 | ManxPower | Carp1: I think they have to enable your account on that server first |
| 22:48.59 | Carp1 | They did |
| 22:49.01 | *** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net) |
| 22:49.04 | Carp1 | Got got the email back. |
| 22:49.06 | h3x | the shadow shido man |
| 22:49.10 | Carp1 | But I don't know the server. lol. |
| 22:49.17 | drooth | do you all recommend installing asterisk with the built in red hat clone? or will any linux flavor do? |
| 22:49.19 | shido6 | whats your username |
| 22:49.21 | ManxPower | MrBelvedr: 64M I think, but I always have 512M in my Asterisk sustems |
| 22:49.36 | Carp1 | carp |
| 22:49.47 | ManxPower | MrBelvedr: The key is that you don't want your system swapping a lot. |
| 22:49.52 | h3x | JerJer[interop]: isn't interop in may? |
| 22:50.14 | shido6 | ok |
| 22:50.40 | shido6 | check your email , carp1 |
| 22:50.42 | *** join/#asterisk linsys (~non@70-57-11-107.dnvr.qwest.net) |
| 22:50.59 | drooth | <PROTECTED> |
| 22:51.03 | ManxPower | gawd that's good |
| 22:51.11 | ManxPower | drooth: any linux |
| 22:51.12 | shido6 | coors light |
| 22:51.15 | tzanger | slackware man... slackware |
| 22:51.15 | h3x | a whole car? thats a helluva lot of guinness |
| 22:51.17 | drooth | thx |
| 22:51.20 | h3x | you will turn green soon |
| 22:51.25 | linsys | Can someone show me a sample extensions.conf that hands all calls off to the local POTS line? I can get calls into my PBX and have all the lines ring, I even got some automated into stuff.. but I can't dail out.. |
| 22:51.33 | ManxPower | shido6: I think the proper term for Coors Light is the term "disgusting" |
| 22:51.37 | sivana | ya, slackware |
| 22:51.42 | sivana | I second that motion |
| 22:51.50 | tzanger | linsys: exten => NXXXXXX,1,Dial(Zap/g1/${EXTEN},,g) |
| 22:51.55 | tzanger | exten => NXXXXXX,2,Hangup |
| 22:52.01 | Darwin[laptop] | debian |
| 22:52.05 | ManxPower | linsys: exten => 91NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN:1}) |
| 22:52.07 | tzanger | (for north america 7-digit local dialing) |
| 22:52.10 | Darwin[laptop] | its cleaner and smaler base |
| 22:52.20 | linsys | I have 10 digit |
| 22:52.21 | ManxPower | mine is for north america 11 digit toll dialing |
| 22:52.22 | tzanger | Darwin[laptop]: how big is your default debian install? |
| 22:52.26 | tzanger | linsys: then adapt the script to work |
| 22:52.35 | Darwin[laptop] | 80 megs |
| 22:52.44 | tzanger | Darwin[laptop]: that's not bad |
| 22:53.02 | sivana | great.. maybe I should have went Debian.. hehe |
| 22:53.34 | tzanger | hahaha My default is around 100M or so |
| 22:53.38 | Darwin[laptop] | brb doorbell |
| 22:53.43 | linsys | I have a line like this include => always-out-pots should i define what you said aboive under the heading [always-out-pots] |
| 22:53.56 | linsys | <shido6> wow |
| 22:53.56 | linsys | <shido6> we can provide you with an asterisk confguration overview for $85 bucks, which is a one hour session |
| 22:53.56 | linsys | <shido6> use greg@nufone.net if we get disconnected |
| 22:54.00 | linsys | what's up with this? |
| 22:54.24 | tzanger | linsys: you are coming across as someone who wants people to solve his problems for him rather than try to learn |
| 22:54.24 | ManxPower | linsys: That is Shido getting tired of your questions 8-) |
| 22:54.34 | *** join/#asterisk FarrisG (~farris@c-24-1-113-24.hsd1.tx.comcast.net) |
| 22:54.40 | ManxPower | I'd charge $120. |
| 22:54.41 | tzanger | shido6 is offering paid support, which is what I was about to do too :-) |
| 22:54.43 | *** join/#asterisk Arauto (~leandro@200141234154.user.veloxzone.com.br) |
| 22:54.51 | ManxPower | People that don't want to config them selves should feel pain. |
| 22:54.54 | fearnor | 85$ is a good deal |
| 22:54.58 | tzanger | it is actually |
| 22:55.06 | FarrisG | Is there a way to configure asterisk so that voicemails are removed from the mailbox after they are emailed? |
| 22:55.06 | fearnor | you should take it. |
| 22:55.12 | tzanger | and nufone's setup is actually pretty robust |
| 22:55.14 | ManxPower | FarrisG: yes |
| 22:55.30 | tzanger | fearnor: damn, allergy season must cost you a fortune |
| 22:55.31 | FarrisG | ManxPower: Is it in the docs somewhere? |
| 22:55.36 | fearnor | hehe |
| 22:56.01 | ManxPower | FarrisG: Did you look in /path/to/asterisk/configs/voicemail.conf.sample |
| 22:56.24 | linsys | Gee.. I didn't know this channel was a bunch of elitists |
| 22:56.43 | fearnor | put it this way |
| 22:56.45 | h3x | Hmmmm |
| 22:56.50 | fearnor | you are providing commercial service to your customers? |
| 22:56.51 | ManxPower | linsys: no. We are capitalists |
| 22:57.08 | h3x | http://www.ds3switch.com/ |
| 22:57.15 | h3x | ^^^ that sure does look like a modified linksys |
| 22:57.17 | linsys | Acutally I've asked nothing but that question which I received any usefull info.. |
| 22:57.30 | ManxPower | Step 1: I do something for you. Step 2: You start handing me piles of cash until I tell you to stop. |
| 22:57.31 | tzanger | it has nothing to do with elitism |
| 22:57.31 | h3x | Is there some easy way they could have hacked a ds3 interface onto it |
| 22:57.34 | linsys | untill that question.. |
| 22:57.36 | ManxPower | See? Quite simple really. |
| 22:57.40 | tzanger | linsys: you aren't even trying to solve your problem |
| 22:57.43 | linsys | Everything else I configured my self.. |
| 22:57.44 | fearnor | h3x: it isn't |
| 22:57.50 | tzanger | I gave you 7 digit dialing and what's the first thing you said "but I need 10" |
| 22:57.51 | fearnor | h3x: just color scheme. |
| 22:57.58 | ManxPower | linsys: There are Wiki pages about this. |
| 22:58.02 | linsys | I had someone look over a config I did... and the box was too slow to work, changed it to a new box, and it all worked |
| 22:58.03 | h3x | have you used one? |
| 22:58.08 | fearnor | reading is FUNDAMENTAL |
| 22:58.11 | fearnor | h3x: no. |
| 22:58.22 | linsys | I've been reading the wiki |
| 22:58.29 | fearnor | ok fine |
| 22:58.34 | fearnor | understanding is also fundamental |
| 22:58.42 | linsys | well I did get this far.. |
| 22:58.56 | tzanger | Darwin[laptop]: my pared down slackware install's about 280M but that includes both perl and python |
| 22:59.02 | linsys | like I said I don't see all the help people here have provided me.. |
| 22:59.13 | linsys | I must have missed those messages.. ??? |
| 22:59.16 | *** join/#asterisk Rick_Hunter (~rhunter@04-120.008.popsite.net) |
| 22:59.22 | *** part/#asterisk mkhan (~mkhan@dsl092-066-137.bos1.dsl.speakeasy.net) |
| 22:59.28 | tzanger | linsys: we've provided a lot of help to a lot of people... some understand and some don't |
| 22:59.32 | fearnor | you must have not been paying attention. |
| 22:59.33 | fearnor | shrug. |
| 22:59.35 | *** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it) |
| 22:59.43 | ManxPower | linsys: The thing is, if you configured everything else, then you should EASILY be able to configure dialing out. |
| 23:00.13 | sivana | :) |
| 23:00.31 | linsys | I was just looking for a URL that;s all, I didn't see anything but a complicated config which was setup to determine long distance calls and send them to a SIP provider and determine local calls and send them to the local POTS.. |
| 23:00.32 | tzanger | JerJer[interop]: that's nothing special... if I tried I get down to 8M or so but that's not a "pared down slackware install" that's a custom linux distro like yours :-) |
| 23:00.46 | linsys | That didn't seem to work for me.. I just couldn't find anything else.. my fault.. |
| 23:00.48 | tzanger | linsys: www.mixdown.ca/~andrew/extensions.conf |
| 23:00.56 | Darwin[laptop] | ahh ok cool |
| 23:00.59 | tzanger | that is a config pulled from a working system |
| 23:01.13 | Darwin[laptop] | well my sister is here back in a bit |
| 23:01.31 | linsys | tzanger: thanks.. |
| 23:01.34 | JerJer[interop] | tzanger: yes...i started with a kernel, glibc and lilo |
| 23:01.41 | JerJer[interop] | then added busybox |
| 23:01.54 | tzanger | JerJer[interop]: yes busybox makes things very small |
| 23:01.55 | h3x | JerJer[interop]: isn't interop in may? |
| 23:01.56 | tzanger | as does uclibc |
| 23:02.11 | tzanger | linsys: that does 7, 10, 11 and international |
| 23:02.14 | JerJer[interop] | h3x: the show is, yes - we are doing the testing now |
| 23:02.21 | h3x | testing what? |
| 23:02.23 | JerJer[interop] | tzanger: yeah i don't like uclibc |
| 23:02.29 | *** join/#asterisk nDuff (~cduffy@net-6621942-66.customer.corenap.com) |
| 23:02.29 | JerJer[interop] | h3x: hot stage |
| 23:02.32 | tzanger | JerJer[interop]: 9M with glibc?? |
| 23:02.34 | h3x | oh |
| 23:02.37 | JerJer[interop] | everything |
| 23:02.38 | *** join/#asterisk Veryhot (Veryhot@adsl-68-125-234-1.dsl.sndg02.pacbell.net) |
| 23:02.43 | JerJer[interop] | tzanger: yes |
| 23:02.46 | h3x | juggling asterisk balls? heh |
| 23:02.47 | tzanger | JerJer[interop]: wow. now that *is* impressive. |
| 23:02.55 | Veryhot | Anyone using Voipjet for Intl ? |
| 23:02.56 | h3x | so you're here in vegas huh |
| 23:03.07 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 23:03.15 | JerJer[interop] | no Belmont |
| 23:03.16 | JerJer[interop] | ca |
| 23:03.20 | JerJer[interop] | ~ San Fran |
| 23:03.26 | Carp1 | I cant get it to work! lol |
| 23:03.27 | h3x | ah i thought interop was in vegas in may? |
| 23:03.46 | JerJer[interop] | it is - but the testing facility is n belmont, ca |
| 23:03.52 | h3x | i see |
| 23:03.57 | tzanger | what is interhop? |
| 23:04.04 | ManxPower | I thought there was more than one Interop show. |
| 23:04.17 | h3x | there is but not in one time of the year |
| 23:04.18 | h3x | heh |
| 23:04.27 | ManxPower | Oh, not that merged with Networld. |
| 23:04.55 | tzanger | Darwin[laptop]: actually pared down slack is 200M with perl and python, I had three complete sets of kernel modules in there |
| 23:05.07 | JerJer[interop] | tzanger: demonistrates interoperability of these so-called open standards we use |
| 23:05.08 | tzanger | but that includes a lot of useful libs |
| 23:05.09 | *** join/#asterisk pol^pht (~pol@adsl-data-63.84-47-32.telecom.sk) |
| 23:05.21 | tzanger | JerJer[interop]: ahh so that is why you've been asking so many SIP questions |
| 23:05.23 | h3x | JerJer[interop]: did they put you in charge of H.323 *snicker* |
| 23:05.25 | Veryhot | im having problem using VoipJet codec for Intl. can someone help me? |
| 23:05.27 | pol^pht | hello all |
| 23:05.31 | JerJer[interop] | SIP |
| 23:05.34 | Carp1 | anyone connected to NuFone through a Budgetone? |
| 23:05.34 | ManxPower | This stupid built in video on my new motherboard does not support the resolution I want. |
| 23:05.55 | h3x | JerJer[interop]: You should just rant and rave and tell everybody to implement IAX2 |
| 23:06.01 | tzanger | h3x: I agree 100% |
| 23:06.05 | h3x | They have a month, now hurry up!@! |
| 23:06.09 | *** join/#asterisk dom-server (Dom@81-86-94-189.dsl.pipex.com) |
| 23:06.31 | dom-server | Is there anyway to link one global to another ? |
| 23:06.37 | *** join/#asterisk iosahdf (~oiashdf@68.71.213-34.atlsfl.adelphia.net) |
| 23:06.38 | dom-server | as in a callerid |
| 23:06.55 | ManxPower | It can do 1024x768 and 2048x1280, but not 1152x864 |
| 23:07.00 | tzanger | heh |
| 23:07.00 | pol^pht | i have "Unable to create channel of type 'SIP'" and no busy tone. is it normal? (asterisk 1.0.5) |
| 23:07.03 | JerJer[interop] | but there are people here testing everything relating to networking |
| 23:07.25 | tzanger | pol^pht: you must have a Hangup right after or something |
| 23:07.40 | dom-server | is there anyway to relay a callerid from 1 users context into the rest of a dial plan? |
| 23:07.50 | ManxPower | pol^pht: PASTE the Dial line. |
| 23:07.57 | ManxPower | dom-server: that is the default. |
| 23:08.11 | Veryhot | I keep getting this msg"channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2" |
| 23:08.30 | pol^pht | ManxPower: exten => 222,1, Dial(SIP/222,40,t) |
| 23:08.33 | h3x | [pr newswire] Las Vegas NV, Networld+Interop. Digium X110P sales boast record growth thanks to Avaya, Nortel, Cisco, 3Com, and Mitel buying large amounts of cards for zaptel timing to support their new IAX2 protocol implementation. |
| 23:08.36 | dom-server | ManxPower : http://stats-box.alpha-networks.co.uk/~files/extensions.conf , im trying to get it so that i can set the callerid in the [user*] contexts |
| 23:08.43 | dom-server | and dont have to add all the other stuff too it |
| 23:08.47 | dom-server | any idea's? |
| 23:09.02 | ManxPower | dom-server: What do you want to set the callerid to? |
| 23:09.05 | fearnor | hahahaha |
| 23:09.08 | pol^pht | tzanger: i need people will hear busy tone |
| 23:09.11 | fearnor | h3x: thats golden |
| 23:09.28 | dom-server | ManxPower : I want it so every user that is added as in [user1] etc can have a differant callerid added obviously |
| 23:09.30 | tzanger | pol^pht: then make your dialplan do that. :-) |
| 23:09.35 | dom-server | becasue they will be dialing out from a differant number |
| 23:09.39 | h3x | hell yeah |
| 23:09.49 | h3x | jerjer would be getting a phat commission check |
| 23:09.50 | ManxPower | dom-server: you don't add users to extensions.conf. You add them to sip.conf, iax.conf, or zaptel.conf. |
| 23:09.54 | tzanger | wow MGM just fucked up I think |
| 23:10.18 | pol^pht | tzanger: that's the question. will asterisk send busy signal if SIP phone not connected? |
| 23:10.19 | ManxPower | dom-server: and where you add the user is where you set their callerid |
| 23:10.20 | tzanger | too bad the DVDA wont' say the same thing |
| 23:10.29 | tzanger | pol^pht: if you configure your dialplan to do so, of course it will |
| 23:10.36 | ManxPower | pol^pht: No it won't unless you set it up to do so. |
| 23:10.39 | dom-server | ManxPower, what do i add to there context in sip.conf? |
| 23:10.53 | ManxPower | pol^pht: see macro-stedexten] in extensions.conf.sample |
| 23:11.03 | ManxPower | dom-server: callerid=robert Dobbs <666] |
| 23:11.07 | ManxPower | ..er. |
| 23:11.11 | ManxPower | dom-server: callerid=Robert Dobbs <666> |
| 23:11.31 | ManxPower | or more accuratly |
| 23:11.43 | ManxPower | dom-server: callerid=George W. Bush <666> |
| 23:11.47 | ubergoober | What should one look at to trouble-shoot one-way connections thru iax? I can receive calls, but I can't initiate them without getting busy respones |
| 23:11.48 | dom-server | But number isnt 666 :S |
| 23:11.58 | pol^pht | ManxPower&tzanger: that you. now i'm sure where to look:) thanks again |
| 23:11.59 | dom-server | Im on about it sending the whole number as in 0845.... |
| 23:12.08 | pol^pht | aah, thank you:) |
| 23:12.12 | ManxPower | dom-server: then set the right number. |
| 23:12.23 | *** join/#asterisk _asr (asr@pimpbox.latency.net) |
| 23:12.26 | dom-server | so in sip.conf |
| 23:12.34 | dom-server | callerid=NAme <0845..> |
| 23:12.34 | dom-server | ? |
| 23:12.37 | nesys | hi folks .. there's someone that could help me to find a mistake about sip trunk between ccme and asterisk, and no call-forward from ccme to asterisk while a simple call from ccme to asterisk works fine? |
| 23:12.41 | ManxPower | dom-server: yes. |
| 23:12.46 | harryvv | seeems voipjet has had alot of 2000ms lags lately. |
| 23:12.54 | dom-server | ManxPower what needs to be in the dialplan for that? |
| 23:12.55 | dom-server | nothing |
| 23:13.04 | ManxPower | Now the telco may or may not accept you setting the callerid. Almost none support NAME, and only some support NUMBER |
| 23:13.16 | nesys | this is my debug: http://www.pastebin.com/266699 |
| 23:13.25 | ManxPower | dom-server: If the SIP user authenticates then it will get the callerid. |
| 23:13.55 | ManxPower | dom-server: How do you know the callerid is not being set? Put a Noop(CALLERID=${CALLERID}) in your dialplan so you can see what callerid is at that point in the dialplan. |
| 23:14.50 | ubergoober | Can anybody point me in the right direction to troubleshoot my iax problem? |
| 23:15.12 | ManxPower | ubergoober: what is your "iax problem"? |
| 23:15.14 | dom-server | So there caller id is set in the sip.conf? |
| 23:15.34 | harryvv | uber is it peer to voip service or peer to peer inside or outside the network |
| 23:15.39 | ManxPower | dom-server: callerid= in sip.conf will override whatever the user sends as their callerid. |
| 23:15.49 | ubergoober | Peer to FWD |
| 23:16.12 | harryvv | peer whats the problem. |
| 23:16.14 | ManxPower | dom-server: I assume you are talking about the Caller*ID of calls made BY the SIP user? |
| 23:16.21 | *** part/#asterisk thomas_adam (~n6tadam@host217-43-99-160.range217-43.btcentralplus.com) |
| 23:16.30 | dom-server | yes |
| 23:16.33 | dom-server | Atm its sending no number |
| 23:16.40 | dom-server | even though i added callerid to sip.conf |
| 23:16.49 | ubergoober | From a FWD registered phone, I can call my own asterisk via the IAX but if I try to dial back the other direction, it's always a busy response |
| 23:16.50 | ManxPower | SECOND POST: dom-server: How do you know the callerid is not being set? Put a Noop(CALLERID=${CALLERID}) in your dialplan so you can see what callerid is at that point in the dialplan. |
| 23:16.54 | ManxPower | I won't ask a 3rd time. |
| 23:17.12 | ManxPower | ubergoober: IAX2 FWD only support ulaw. |
| 23:17.21 | ubergoober | I'm allowing ulaw |
| 23:17.32 | ManxPower | ubergoober: where? |
| 23:17.39 | harryvv | so you are calling that fwd from another phone like a cell to call back into your box ? |
| 23:18.20 | ubergoober | in iax.conf |
| 23:18.20 | dom-server | Its not displaying a number ManxPower |
| 23:18.20 | ManxPower | ubergoober: general or your fwd entry? |
| 23:18.20 | dom-server | Whys that when its set in sip.conf |
| 23:18.20 | ManxPower | dom-server: Until you put the Noop in your dialplan I cannot help you further. |
| 23:18.20 | ubergoober | general |
| 23:18.21 | *** join/#asterisk HaKim (~kaardelen@203.221.251.99) |
| 23:18.31 | harryvv | I never like the idea of FWD to much a risk somone listening in on your convo ;) |
| 23:19.24 | dom-server | That stops it working totally |
| 23:19.29 | dom-server | Where is the best place to put it |
| 23:19.33 | ManxPower | dom-server: then you are doing something wrong. |
| 23:19.37 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 23:19.39 | ManxPower | dom-server: right before the Dial line. |
| 23:19.44 | nesys | Could you help me to understand the debug? |
| 23:20.49 | dom-server | What should the output look like |
| 23:21.35 | *** part/#asterisk FarrisG (~farris@c-24-1-113-24.hsd1.tx.comcast.net) |
| 23:22.08 | ManxPower | dom-server: it should look like -- Executing NoOp("SIP/2121a-5a0b", "CALLERID=Manx Power <2121>") |
| 23:22.13 | ariel_ | harryvv, too much of a risk? someone can listen to any pots line as well. |
| 23:22.53 | dom-server | <PROTECTED> |
| 23:22.54 | h3x | dosent fwd do a transfer |
| 23:23.19 | ManxPower | dom-server: and you set the callerid= line in the [dom] section of sip.conf? |
| 23:23.26 | iosahdf | is there a vonage-like service provider for +1 who supports asterisk? |
| 23:23.27 | ManxPower | Paste the ACTUAL noop line in your extensions.conf. |
| 23:23.30 | dom-server | yes |
| 23:23.39 | dom-server | context=user1 |
| 23:23.39 | dom-server | callerid=Dom Eves <08452413210> |
| 23:23.41 | ariel_ | iosahdf, lots |
| 23:23.48 | ManxPower | Paste the ACTUAL noop line in your extensions.conf. |
| 23:23.58 | iosahdf | ariel_, which are the better ones? |
| 23:24.04 | dom-server | exten => _0Z.,4,Noop(CALLERID=${CALLERID}) |
| 23:24.25 | iosahdf | i'm looking at iaxprovider.net and there's a huuuge list |
| 23:24.31 | Veryhot | anyone using VoipJet for Intl call? |
| 23:24.41 | ManxPower | dom-server: put your sip.conf on pastebin.ca (sans password/secret) |
| 23:24.51 | dom-server | sans? |
| 23:24.57 | ManxPower | sans == without |
| 23:25.04 | ariel_ | better well everyone has there issues including Vonage. I have been using race.com, voicepulse on my system for inbound did. outbound I use voipjet, race.com and nufone. |
| 23:25.12 | ManxPower | "Sans Serif" = "Without Serifs" |
| 23:25.52 | ManxPower | I've been linking teliax these days |
| 23:25.52 | dom-server | http://pastebin.ca/8749 |
| 23:25.52 | ManxPower | VoipJet I won't use. |
| 23:25.59 | Veryhot | max: for Intl? |
| 23:26.09 | Veryhot | max: cheap 1.3 mins |
| 23:26.14 | harryvv | arial I am totally aware of that. My telco instructos said its a requirments in the old days to listen in between two called parties and wait before thay can disconect before thay can switch over the drop cable. Normally this was in the wee hours of the morning. Thay were obviosly strictly fobiden to talk to anyone about what thay hear even if its a crime that is about to happen. But voip is unregulated and thus I dont think that p |
| 23:26.14 | harryvv | eople on FWD would adhear to those same rules. |
| 23:26.16 | Veryhot | max: and good rate on Intl |
| 23:26.38 | Veryhot | manx: who ya using? |
| 23:26.40 | iosahdf | interesting |
| 23:26.57 | ManxPower | Veryhot: I've been linking teliax these days |
| 23:27.04 | ManxPower | .er...likeing |
| 23:27.21 | harryvv | When he said wee hours I did not take the course for the next quarter :) Did not like the idea of staying up between 1-4 am :) |
| 23:27.24 | ManxPower | dom-server: It should be working. |
| 23:27.48 | dom-server | Hmm |
| 23:27.55 | dom-server | do you need to see the extensions file/ |
| 23:28.07 | ManxPower | take out everything except the [general] stuff and the [dom] stuff. |
| 23:28.25 | Veryhot | manx: teliax.com ? |
| 23:28.25 | dom-server | from where? |
| 23:28.30 | ManxPower | dom-server: No. Callerid is not set in extensions.conf unless you want to override something. |
| 23:28.45 | dom-server | [user1] |
| 23:28.45 | dom-server | exten => _*,1,SetCallerID(448452413210) |
| 23:28.49 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 23:28.50 | dom-server | I have things like that in extensions |
| 23:28.56 | harryvv | how is the reliability of teliax? where does it terminate at? |
| 23:29.13 | Veryhot | manxpower: not bad Intl rates |
| 23:29.16 | ManxPower | dom-server: http://pastebin.ca/8750 |
| 23:29.40 | ManxPower | Veryhot: rates are less important than features, DID rate centers, service, quality, etc. |
| 23:29.45 | dom-server | Then what ? |
| 23:29.48 | ManxPower | I do like $5 of toll calling a month. |
| 23:30.06 | ManxPower | dom-server: that is the sip.conf I am suggesting you try. you are not a native english speaker, are you? |
| 23:30.10 | harryvv | I perfer quality over anything else. |
| 23:30.31 | ManxPower | harryvv: quality doesn't matter if you can't get a DID in the city you want. |
| 23:30.32 | Veryhot | manx: yeah, I do need DID |
| 23:30.49 | harryvv | Manx hehe true. I hear wait times are like a month or more? |
| 23:30.50 | dom-server | rofl yes |
| 23:30.52 | h3x | We're going to launch a comprehensive DID coverage soon |
| 23:30.55 | ManxPower | The ONLY reason I'm transitioning away from NuFone is their lack of DIDs |
| 23:31.06 | h3x | www.carrierone.net/dids |
| 23:31.11 | ariel_ | what is this softcap that teliax has on there plans? |
| 23:31.22 | dom-server | Executing NoOp("SIP/dom-8528", "CALLERID=") in new stack |
| 23:31.22 | h3x | i guess it dosent really list rate centers on there, but ask me some :P |
| 23:31.29 | ManxPower | ariel_: I think it means "if the average over three months is more than X we'll spank your ass" |
| 23:31.48 | h3x | I'm even going to have rural america and alaska/hawaii for instance |
| 23:31.50 | ManxPower | ariel_: they explain it on their site somewhere. |
| 23:31.50 | Veryhot | h3x: what's the price for DID? |
| 23:31.55 | *** join/#asterisk zack (~zack@sebastian.redhat.com) |
| 23:31.55 | dom-server | ManxPower : Executing NoOp("SIP/dom-8528", "CALLERID=") in new stack |
| 23:32.01 | h3x | Veryhot: it depends on the underlying carrier |
| 23:32.03 | file[laptop] | dejavu |
| 23:32.10 | ManxPower | dom-server: you ARE doing a RELOAD at the Asterisk CLI, right? |
| 23:32.16 | dom-server | yes |
| 23:32.19 | zack | HaKim: what's with the spam? |
| 23:32.25 | h3x | some of them will be flat rate, others are metered |
| 23:32.29 | ManxPower | dom-server: then I have no more suggestions |
| 23:32.35 | h3x | depends on the rate center |
| 23:32.36 | Veryhot | h3x: do you have any for 858? |
| 23:32.46 | h3x | exchange? |
| 23:32.49 | dom-server | ok Thanks |
| 23:32.58 | zack | is anyone else aware that this HaKim guy is privmsg-spamming on join? |
| 23:33.32 | iosahdf | hm. callid is useful for forwarding to cells :) |
| 23:33.37 | ManxPower | Guinness is proof that god loves us. |
| 23:33.37 | Veryhot | h3x: area code |
| 23:33.53 | h3x | No i mean what exchange |
| 23:34.01 | h3x | or is it all local to each other |
| 23:34.09 | h3x | City i guess |
| 23:34.21 | h3x | hehe i was born in la jolla |
| 23:34.30 | Veryhot | h3x: local yes. |
| 23:34.47 | Veryhot | h3x: I need DID for incoming |
| 23:34.50 | ariel_ | zack, at least I was not the only one to get it. |
| 23:35.09 | harryvv | BTW by default the windows xp built in firewall would block rtp 2 way voice transmissions? |
| 23:35.13 | zack | ariel_: maybe this person should be forcefully ejected. |
| 23:35.14 | h3x | it appears ill have 4 different underlying carriers serving that particular area code |
| 23:35.16 | Veryhot | manxpower: what's this mean for Pay as you go? "A two cents connection charge applies unless call is from another teliax user " |
| 23:35.33 | zack | bkw_: you're the least-idle operator... we have a spammer in here. |
| 23:35.37 | Veryhot | manxpower: so 0.02/min + .02 connection ? |
| 23:35.58 | h3x | level3, mci, pacwest, and oh shit theres two more |
| 23:36.12 | h3x | I'll have global crossing up the soonest of all those probably |
| 23:36.13 | ManxPower | Veryhot: I would assume so. |
| 23:36.28 | ManxPower | h3x: Make sure users can get the info they need from your web site. |
| 23:36.37 | ManxPower | I just tried to open a ticket and it didn't work. |
| 23:36.39 | h3x | Veryhot: I can do a $5/mo flat rate did |
| 23:36.46 | h3x | ManxPower: I know, that is all on the staging server |
| 23:36.47 | Veryhot | h3x: oh nice |
| 23:37.01 | Veryhot | h3x: using iax or? |
| 23:37.03 | h3x | we haven't officially launched it |
| 23:37.03 | zack | anyway. would it be possible to use chan_bluetooth without rebuilding asterisk? |
| 23:37.10 | ManxPower | h3x: Well let us know when you are actually selling service. 8-) |
| 23:37.29 | h3x | Veryhot: we can do SIP by transferring your the RTP directly from the underlying carrier, or we can do IAX |
| 23:37.47 | h3x | ManxPower: I'm selling it, its just not generally released yet |
| 23:37.48 | ariel_ | h3x, what is the co. name? website? |
| 23:37.53 | h3x | www.carrierone.net |
| 23:38.06 | Veryhot | h3x: how can we get a demo ? |
| 23:38.10 | harryvv | cool domain name |
| 23:38.10 | *** join/#asterisk FxMulder (rog@209.159.235.241) |
| 23:38.19 | h3x | Here is how we're dealing with DIDs. We have them going to a SER proxy since most of the carriers have SIP already |
| 23:38.27 | harryvv | h3x when did you register it |
| 23:38.32 | FxMulder | HaKim is onjoining |
| 23:38.54 | zack | FxMulder: yeah, i know. i told an op, waiting for a response :/ |
| 23:38.55 | h3x | some of the providers we have a direct VoIP only IP connection with, others are public internet |
| 23:38.56 | FxMulder | and unless asterisk has gotten into the porn business, I don't think its topic worthy |
| 23:39.04 | h3x | which is fine because it eventually hits the internet anyway :P |
| 23:39.09 | h3x | harryvv: Umm i think 2003 ? |
| 23:39.31 | ManxPower | I want my calls going over the internet as little as possible. |
| 23:39.33 | h3x | by proxying through SER, things like T.38 fax etc will work if you can support it |
| 23:39.39 | harryvv | h3x that was probebly a good time seems any domain name closely related to telco terms all used up for .com. |
| 23:39.51 | h3x | if you want IAX then we just transfer it to our asterisk box and send it with IAX |
| 23:40.21 | Veryhot | h3x: what's your rate for US calls? |
| 23:40.27 | FxMulder | there any good documentation on the integration of asterisk and sphinx? I remember I looked into it a year ago and it was docs were just spawning, I can't find much progress though |
| 23:40.30 | h3x | ManxPower: agreed, we have direct connections to several of them but theres so many carriers for DIDs it would take forever to order private line to them all |
| 23:41.00 | Chuji | FxMulder : still pretty non-existant |
| 23:41.05 | h3x | Veryhot: we do mostly wholesale stuff, so our rate schedules are complex. I really doubt we will ever beat something like voipjet thats practically giving it away |
| 23:41.09 | h3x | in retail space |
| 23:41.12 | iosahdf | is there a provider who offers el-cheapo extensions rather than unique phone numbers |
| 23:41.34 | ManxPower | iosahdf: There are "carriers" that offer FREE extensions. |
| 23:41.38 | harryvv | h3x thay are also having problems. I see alot of 2000ms lag rates from voipjet |
| 23:41.38 | Veryhot | h3x: but you can do incoming DID flat? |
| 23:41.45 | ManxPower | FWD is one and they have a PSTN gateway. |
| 23:41.54 | h3x | One marketing gimmick we are going to have is giving away free DIDs in 10 markets |
| 23:41.57 | iosahdf | nice |
| 23:42.07 | Veryhot | nic |
| 23:42.07 | h3x | the catch is you only get 2 simultaenous calls per DID |
| 23:42.18 | Veryhot | h3x: that's ok :) |
| 23:42.24 | h3x | we can do this because we actually get paid something to pick up calls from those markets heh |
| 23:42.26 | harryvv | interesting |
| 23:42.41 | Veryhot | h3x: I can always use diff provider for outgoing. |
| 23:42.51 | h3x | Veryhot: yeah you can ... |
| 23:43.16 | Veryhot | h3x: I have clients that need unlimited incomind DID |
| 23:43.17 | ManxPower | h3x: I thought all carriers paid each other for terminating each other's traffic. |
| 23:43.19 | h3x | I just don't want to advertise or sell stuff like .013 a minute when I have call center customers doing several T1s worth of VoIP with me at .02/Min :P |
| 23:43.26 | *** join/#asterisk d-tech (~dtc@node-423a1ebb.cle.onnet.us.uu.net) |
| 23:43.47 | h3x | ManxPower: well most of these guys selling wholesale DIDs in their market footprint are making money on both ends |
| 23:44.19 | h3x | manx/veryhot: the problem with "unlimited" is most of them meter something between .003/Min and .009/Min for major markets |
| 23:44.50 | iosahdf | veryhot, what's the gimmick with FWD |
| 23:44.50 | h3x | they charge their customer (me, you, whatever) and the IXC or LEC that dropped that call off on the tandem and collect on both sides |
| 23:44.59 | iosahdf | i mean manxpower |
| 23:45.39 | Veryhot | h3x: when can we signup for one of those flat DID? |
| 23:45.58 | h3x | it'll be a couple more weeks or so |
| 23:46.16 | h3x | one of my carriers services like 98% of california |
| 23:46.17 | Veryhot | h3x; I will cancel my VP DID |
| 23:46.30 | h3x | but the catch is i have to colo with them in cali |
| 23:46.36 | h3x | they send it to me with TDM |
| 23:46.43 | h3x | it also comes in on 3 different trunkgroups |
| 23:47.04 | Veryhot | h3x: colo in LA? |
| 23:47.07 | *** join/#asterisk M-A-D-O-N-N- (~isabel_so@n219078201137.netvigator.com) |
| 23:47.08 | h3x | noley, stockton |
| 23:47.10 | h3x | erfkjfhsdakjfhsda |
| 23:47.12 | h3x | s/noley/no/ |
| 23:47.16 | *** join/#asterisk mog_home (~mogorman@146.229.191.117) |
| 23:47.19 | shmaltz | msg jbot seen shido6 |
| 23:47.36 | shmaltz | sorry guys forgot to / |
| 23:47.36 | h3x | I'll have 858 before then though |
| 23:47.37 | *** part/#asterisk omelia (~jana_009@pc-66-208-83-200.cm.vtr.net) |
| 23:48.05 | h3x | theres no merit for me to get global crossing DIDs over a private connection so that should be up really soon |
| 23:48.16 | Veryhot | h3x: can I get a contact info |
| 23:48.33 | h3x | the larger players like level3, qwest, global, etc. have their session border controllers attached to the core of their network |
| 23:48.37 | *** join/#asterisk maxclan1970 (haber_si_l@119.Red-81-38-170.pooles.rima-tde.net) |
| 23:48.43 | Veryhot | h3x: or signup in the DID page? |
| 23:48.46 | h3x | so there aint much of a point in messing around with that stuff |
| 23:49.04 | h3x | just use the For More Information... thing, it will email me or somebody else that can help you |
| 23:49.20 | Veryhot | h3x: ok, just mention $5 DID flat? |
| 23:49.20 | h3x | if you do it on the page that is of particular interest that helps |
| 23:49.31 | h3x | yeah, and do you need to port a number |
| 23:49.42 | h3x | I immediately have 702/777-XXXX numbers in vegas |
| 23:49.42 | h3x | heh |
| 23:50.21 | mog_home | there are spam-bots in #asterisk |
| 23:50.28 | mog_home | or now just one |
| 23:50.57 | h3x | One of the other reasons im not really too crazy about selling outbound to people that get inbound from me is that generally all of your "local" calls are going to be an expensive intrastate call for me |
| 23:51.06 | h3x | until i get some more 1+ providers in that don't have this problem |
| 23:51.37 | Veryhot | h3x: sent. |
| 23:51.46 | h3x | ok cool |
| 23:52.02 | h3x | Veryhot: do you need to port a number or just get a new one? |
| 23:52.44 | *** part/#asterisk maxclan1970 (haber_si_l@119.Red-81-38-170.pooles.rima-tde.net) |
| 23:54.13 | h3x | of course our main objective is to sell these things wholesale anyway |
| 23:54.16 | *** mode/#asterisk [+r] by bkw_ |
| 23:54.32 | bkw_ | I took +r off for testing to see if the spam bots would go away |
| 23:54.34 | bkw_ | but they are still out |
| 23:54.37 | bkw_ | +r back |
| 23:54.49 | h3x | we're working on a provisioning interface for ssl tcp socket, and web extranet, so that emerging voip providers can assign new numbers out of our pools, do local number portability, etc. all in one unified interface |
| 23:54.53 | iosahdf | freegin robots |
| 23:55.31 | h3x | do a full LIDB dip to find out who the current LEC is |
| 23:55.38 | h3x | and other information to scrub the account |
| 23:55.42 | h3x | all kinds of backoffice features |
| 23:57.31 | Veryhot | nice |
