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| 00:15.05 | grEvenX | who's running the asterisk.org webteam? |
| 00:15.05 | grEvenX | :P |
| 00:15.20 | grEvenX | http://www.asterisk.org/feedback is full of spam |
| 00:15.40 | grEvenX | which leaves me to think there's probably no use in posting anything there |
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| 00:26.26 | rdircio | yea in that feedback page they got: ....Low cost dental Plans.... |
| 00:26.28 | rdircio | lol |
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| 00:56.29 | hardwire | http://hardwire.pastey.net/135198 |
| 00:56.30 | hardwire | woot |
| 00:56.57 | hardwire | map user=phone and isup-oli into from-user=phone and from-isup-oli=xx |
| 00:57.34 | rdircio | ?... o..k... |
| 00:59.32 | hardwire | it's important |
| 00:59.34 | hardwire | truuust me |
| 00:59.47 | rdircio | ok my outsourced friend |
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| 01:12.04 | hardwire | I suppose it's easy enough to just check to see if isup-oli=70 is in CHANNEL(from) |
| 01:12.09 | hardwire | but it's not as fun in any way |
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| 01:19.39 | *** join/#asterisk catlynx (sione@ocs.net) |
| 01:20.07 | catlynx | I just got a TDM410 for my box, anyone know where I can find sample config on how to set this up? |
| 01:20.36 | catlynx | it has 2 FXO and 2 FXS |
| 01:22.24 | hardwire | I'll send you instructions if you send it to me. |
| 01:22.29 | hardwire | -> home |
| 01:22.39 | catlynx | :P |
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| 01:23.16 | catlynx | im getting lost on google trying to find docs on it |
| 01:23.16 | hardwire | catlynx: you with vmware? |
| 01:23.40 | catlynx | hardwire: nope |
| 01:24.20 | catlynx | hardwire: why you ask? |
| 01:24.27 | hardwire | nm.. thats n6il |
| 01:24.39 | hardwire | you don't have a web page.. bad. |
| 01:24.41 | catlynx | yup Mikey |
| 01:24.57 | catlynx | I have a web page just this not my box :P |
| 01:25.13 | hardwire | oh well.. I'm headed home. |
| 01:25.44 | hardwire | catlynx: do you know about ~book? |
| 01:26.06 | catlynx | nope |
| 01:26.09 | hardwire | ~book |
| 01:26.10 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:26.16 | catlynx | oh that book |
| 01:26.24 | catlynx | it covers TDM410? |
| 01:26.36 | hardwire | I believe it as well as the asterisk sample configs has samples and ample docs |
| 01:26.37 | hardwire | if you wanna dig in |
| 01:26.47 | catlynx | i will check that |
| 01:26.49 | hardwire | the dahdi docs combined with the asterisk docs will = working TDM410 |
| 01:26.57 | hardwire | bbl |
| 01:27.02 | catlynx | heh thanks |
| 01:27.38 | hardwire | also.. I believe digium provides samples |
| 01:27.41 | hardwire | mmmmmmmm |
| 01:27.43 | hardwire | here |
| 01:27.44 | hardwire | https://www.digium.com/en/supportcenter/documentation/viewdocs/TDM410 |
| 01:28.14 | catlynx | ah sweet thats what i needed :) |
| 01:28.18 | catlynx | thanks |
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| 01:43.37 | Devon_ | ariel_ |
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| 01:58.20 | TJ^ | anyone know how good the Linksys SPA-3000 is? |
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| 01:59.19 | grinder13 | hello! |
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| 02:00.56 | grinder13 | i am using asterisk 1.6.2.6 on a CentOS server. does it support zrtp? |
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| 02:31.55 | manxpower | grinder13: do you have any indication Asterisk supports zrtp on ANY OS? |
| 02:34.48 | grinder13 | i know that phil zimmerman provides a patch: http://zfoneproject.com/prod_asterisk.html |
| 02:35.17 | *** part/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 02:35.24 | grinder13 | there is also this bug: https://issues.asterisk.org/view.php?id=10024 |
| 02:35.27 | manxpower | try checking the changelogs and UPGRADE*.txt in your Asterisk source tarball |
| 02:35.47 | grinder13 | yes, i know |
| 02:35.57 | grinder13 | have to check again though |
| 02:36.34 | grinder13 | the think is if it is included in the rpm packages provided by digium's repos for centos |
| 02:38.47 | grinder13 | also do you know if srtp is included? or do you have to compile asterisk in order to have it |
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| 02:39.10 | grinder13 | i am asking because of that: http://forums.digium.com/viewtopic.php?f=13&t=73717&p=143653&hilit=srtp&sid=0302fae0d330001b400a110a4663d341#p143653 |
| 02:42.10 | grinder13 | also regarding h323, does asterisk has support for h235 encryption? |
| 02:42.18 | grinder13 | i cant find anything on that |
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| 05:38.43 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2, 1.6.0.4, 1.4.10 (2009/12/02), dahdi-linux 2.2.1 + dahdi-tools 2.2.1 (2010/01/20), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
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| 06:08.40 | spenguin[work] | is there a page where we can figure the dialplan syntax,cmd differences between asterisk versions? |
| 06:09.17 | spenguin[work] | something like that would be nice |
| 06:16.43 | TrippaOP | Hi, how do i change the default email that asterisk sends out, i.e. when new emails voice messages arrive? |
| 06:25.41 | ChannelZ | it's all in voicemail.conf |
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| 07:35.48 | Jumpie | i nkow there are a slew of sites probably, but is there an officially recomended site for locking down your box? |
| 07:35.53 | Jumpie | security wise, common vulnerabilities, etc |
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| 08:51.34 | yang | I get this sipsak message |
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| 08:51.38 | yang | sipsak -vv -s sip:026203201@IP |
| 08:51.41 | yang | received ICMP message (type: 3, code: 3): from IP |
| 08:51.44 | yang | what does it mean ? |
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| 09:08.47 | sawgood | Any tips for when I call a DID (which is set to play an IVR file) (system recording) ... the line plays only dead air? |
| 09:08.59 | sawgood | But when I listen to the .wav file from the box ... it plays just fine? |
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| 09:28.19 | kruemeltee | hello all together :-) |
| 09:36.06 | kruemeltee | sometimes it happens, that the caller has disabled his callerid ... is it possible to extract the callerid anyhow? |
| 09:36.22 | kruemeltee | mybe from the sip-privacy header or something like this? |
| 09:37.21 | kruemeltee | brb ... |
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| 09:44.15 | jozu | hi to all |
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| 09:54.12 | kruemeltee | hi jozu |
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| 10:13.17 | kruemeltee | is there any possibility to extract the complete SIP Header during a call? |
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| 10:20.51 | spenguin[work] | sip debug? |
| 10:21.05 | spenguin[work] | or plain tcpdump |
| 10:22.34 | kruemeltee | tcpdumb is too much information ... I think ;-) what about sip debug? just entering within the CLI console? |
| 10:23.42 | kruemeltee | question is, if sometimes the hidden CALLERID is somewhere within the header |
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| 11:02.51 | mutax | can someone help me getting misdn compiled? http://pastebin.com/CcZfhQ9q Is this a kernel, userland, gentoo, debian or general fuckup? |
| 11:03.35 | mutax | besides the fact that I hat to edit the Makefile for 2.6 kernel, because it changed CFLAGS directly -.- |
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| 11:04.20 | mutax | is a bit annoyed because neither in gentoo nor debian there seems to be any working support for this right now.. |
| 11:06.12 | WIMPy | mutax: That looks rather old. What kernel ar you using? The old misdn isn;t supported any more. |
| 11:08.39 | mutax | so http://www.misdn.org/index.php/MISDN_v2_Download is not needed any more? |
| 11:08.49 | Takapa | hi, any recommendations on an ata with many fxs ports? |
| 11:09.01 | mutax | WIMPy: do you know of a recent howto then ;) |
| 11:09.05 | mutax | kernel is 2.6.33 |
| 11:09.18 | Cresl1n | mutax, any reason to use misdn instead of DAHDI + libpri? |
| 11:09.19 | Takapa | i need ~35, but two atas with 20 each would be okay.. |
| 11:09.22 | WIMPy | mutax: misdn2 is in the standard kernel now. You only need misdn-user. |
| 11:09.32 | mutax | ahh. ok... |
| 11:09.45 | mutax | Now I read it there on the page :) |
| 11:09.52 | mutax | *grr* :) |
| 11:10.24 | WIMPy | Cresl1n: That usually needs patching. But some time I will try it :-) |
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| 11:34.56 | angryuser | what what the setting which forbid the pri channel reset on asterisk start (in zapata.conf) ? |
| 11:35.23 | angryuser | resetinterval |
| 11:36.46 | mutax | WIMPy: the people at misdn.org don't want errors in the wiki to be corrected, or why can't I make an account there to edit pages? |
| 11:40.46 | WIMPy | mutax: nfi |
| 11:41.14 | WIMPy | I also failed to find documentation on the APIs. |
| 11:42.29 | mutax | installs asterisk from source now - distributors seem to fail :) |
| 11:42.36 | mutax | the git version of misdn-user works |
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| 11:45.28 | Belgarath | hi |
| 11:45.51 | Belgarath | Failed to authenticate on INVITE to I got that message evry tim I try to make outgoing sip call after upgrade from 1.4 to 1.6 |
| 11:47.22 | TJ^ | whats the biggest setup u guys know of/have seen/have? |
| 11:48.35 | Cresl1n | WIMPy: shouldn't need patching anymore, since DAHDI and libpri support BRI? |
| 11:49.25 | Cresl1n | (was wondering because I was the one that wrote a lot of the libpri and DAHDI support for BRI :-) ) |
| 11:49.55 | WIMPy | Cresl1n: At least no support for NT-Mode on an octobri out of the box. |
| 11:50.23 | Cresl1n | what's the patch for then? |
| 11:52.27 | WIMPy | Cresl1n: To be able to switch to NT Mode via module parameters. Don't know it that's all. I didn't try it. |
| 11:52.59 | Cresl1n | I think you can do that with the wcb4xxp driver (IIRC) |
| 11:53.13 | Cresl1n | oh way, you have to change the jumpers too |
| 11:53.43 | WIMPy | Cresl1n: That was for the wcb4xxp, yes. |
| 11:54.34 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
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| 11:57.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:58.49 | Chainsaw | Morning Leif. |
| 11:59.15 | leifmadsen | morn! |
| 12:06.56 | suprstar | http://www.cdwg.com/shop/products/default.aspx?EDC=1409443 - can I configure this to be a single dahdi channel, ie will dahdi recognize it as legit comm hw and let me use it? |
| 12:07.24 | Belgarath | what does sip 404 actualy means ? |
| 12:07.33 | Belgarath | is it that the number doe s not exist |
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| 12:07.36 | Belgarath | or sth else ? |
| 12:08.10 | WIMPy | suprstar: no |
| 12:08.17 | WIMPy | Belgarath: No text? |
| 12:08.32 | suprstar | 404 Not Found: Sent from the proxy when either the intended party does not exist (wrong number dialed) or the user is not registered on that proxy. This message might be sent from the gateway if it's caused by dialing a wrong number. |
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| 12:09.31 | *** mode/#asterisk [+o file] by ChanServ |
| 12:11.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:11.24 | suprstar | wimpy: what's the problem with that modem? I have another server with a te410p and some unknown modem - the modem shows up in zttool as "Wildcard X101P Board 1" - i need something similar on my new server |
| 12:13.00 | WIMPy | Woot? Is it something special or does dahdi now support what used to be chan_modem? I thought that was abandoned? |
| 12:13.26 | WIMPy | Without replacement, that is. |
| 12:14.38 | suprstar | the old system is zap, not dahdi, and i didnt set it up.. the guy who did could very well have hacked up the source to make it work. |
| 12:15.21 | suprstar | no docs whatsoever, and he quit, disgruntled, will not help us... |
| 12:16.24 | Belgarath | WIMPy: i cannot dial out |
| 12:16.44 | Belgarath | whenever I dial out over sip in *1.6 i got 404 and congestion |
| 12:16.56 | Belgarath | sip show registry show me registered with providers |
| 12:17.05 | [TK]D-Fender | Belgarath: 404 = your number is bad |
| 12:17.35 | [TK]D-Fender | Belgarath: and this issue has nothing to do with registration status. the number is bad |
| 12:17.56 | *** join/#asterisk redax (redax@r6.hu) |
| 12:17.57 | suprstar | not YOUR reg. status anyway.. |
| 12:17.58 | redax | hi. |
| 12:18.00 | [TK]D-Fender | Belgarath: So go look at your call |
| 12:18.28 | Belgarath | [TK]D-Fender: it was working before upgrade to 1.6 with same details |
| 12:18.48 | Belgarath | and the numbe I am calling is mine :) |
| 12:18.49 | [TK]D-Fender | Belgarath: Stories don't help... output & configs do. |
| 12:21.54 | redax | why does my quadbri changes the timing source time to time from port 1-4 if I specified the spans like span=1,<PORTNUM>,0,css,ami ? this way port1 should be the first timing source and port2 the secondary ... etc.. |
| 12:23.18 | mutax | WIMPy: do you have a minute for me? |
| 12:23.36 | WIMPy | yes |
| 12:23.40 | mutax | great :) |
| 12:24.05 | mutax | I am trying to get asterisk with the usdb isdn adapter going. the adapter is btw. as you said the same you have |
| 12:24.28 | mutax | misdn is in the kernel. misdn-tools from git are working, misd_info shows the adapter |
| 12:24.45 | mutax | the misdn-user is installed in /usr/local/ |
| 12:25.01 | mutax | now I try to compile asterisk, but the configure script does not find misdn |
| 12:25.21 | mutax | asterisk is 1.6 vanilla |
| 12:25.41 | WIMPy | No, chan_misdn only worked with the old misdn. You need LCR in between. |
| 12:26.00 | WIMPy | http://www.linux-call-router.de/ |
| 12:26.16 | mutax | *uff* okay ;) Is this true for asterisk 1.4 too? |
| 12:26.58 | WIMPy | .me only tried 1.6 |
| 12:27.03 | mutax | k |
| 12:27.13 | mutax | thanks ye again... what a mess :D |
| 12:27.14 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:28.10 | WIMPy | Yes, but maybe we will be able to get some basic phone features working that way some day. |
| 12:31.00 | WIMPy | So far, I think parking is the only thing that works, but that's more than I have senn from any other solution :-( |
| 12:31.15 | mutax | heh |
| 12:31.25 | mutax | Hmm. Do I have to cross link the ports, because the card receives what the telephone sends and vice versa." |
| 12:32.00 | WIMPy | If you want to connect a phone, you need a crossover cable, yes. |
| 12:32.22 | WIMPy | And a power source for the phone. |
| 12:32.23 | mutax | i like to connect the usb adapter as a phone |
| 12:33.00 | mutax | it says its a isdn TA so this should'nt be neccessary, right? |
| 12:33.12 | WIMPy | In that case any straigt cable will do. |
| 12:33.25 | mutax | k |
| 12:33.33 | mutax | Oh great - I just got a reply from sipgate |
| 12:33.57 | mutax | they say setting the callerid does not work in the basic-account with asterisk |
| 12:34.11 | mutax | only woth two pbx by siemens and avm |
| 12:34.25 | mutax | *narf* 4 hours of my life wasted... |
| 12:35.45 | [TK]D-Fender | I doubt that. |
| 12:36.41 | WIMPy | Sounds pretty strange, yes. What's that about? |
| 12:39.52 | suprstar | Mu$t be $ome rea$on.. Are the re$eller$ of $iemen$ or avm by any chance? |
| 12:40.56 | mutax | WIMPy: I don't know. I just asked them if they woul tell me what these PBXes do on the sip channel |
| 12:41.17 | mutax | suprstar: they are solely a sip provider |
| 12:41.39 | mutax | I kept getting 403's on the sip channel there (check from field) |
| 12:41.50 | mutax | but in the past I was able to set the id from ekiga |
| 12:42.13 | mutax | and 2 years ago their support wrote me what to do in asterisk to set the callerid! |
| 12:42.14 | [TK]D-Fender | mutax: then you should probably be checking your FROM field. |
| 12:42.20 | mutax | :D |
| 12:42.27 | mutax | yes, well, no |
| 12:42.56 | mutax | [TK]D-Fender: the from field was exactly the field one of theie technicians wrote me in a mail on how to set the callerid with them |
| 12:43.13 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-oaedeclegpdlxfum) |
| 12:43.47 | [TK]D-Fender | mutax: ok, so whats the problem? Go look at the SIP debug of yoru call attempts |
| 12:43.51 | mutax | I guess they implemented a filter for the cheap accounts and the two PBXs somehow circumwent this or are otherwise broken. Maybe I should change my user-agent :) |
| 12:44.15 | [TK]D-Fender | mutax: Maybe you did something wrong. Aside from developing an aversion to looking. |
| 12:44.35 | mutax | [TK]D-Fender: I prefer wireshark, but yes, thats basically what I do. the sip dialog is okay. |
| 12:44.50 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 12:45.09 | mutax | [TK]D-Fender: but they seem to throw the callid in the from field away and replace it by my sipgate number |
| 12:45.50 | pabelanger | Has anybody seen any audio delay from a Aastra 6731i? IE: Asterisk calls the Aastra, answer the call, 5+ seconds before audio is bridge? Asterisk 1.4.25 |
| 12:46.59 | mutax | WIMPy: ehrm.. just to get that right. You told me, I need LCR to connect an asterisk to my ISDN coming on from my provider? |
| 12:47.03 | [TK]D-Fender | mutax: "sip dialog is okay" <- You might a second, or thirty-second set of eyes on that |
| 12:47.08 | *** join/#asterisk Faithful (~Faithful@119.12.29.20) |
| 12:47.37 | mutax | [TK]D-Fender: I'll see if I find one and pastebin it ... |
| 12:47.53 | WIMPy | mutax: yes, that's with misdn2. |
| 12:48.39 | *** join/#asterisk rdircio (~admin@201.137.57.68) |
| 12:48.39 | mutax | WIMPy: okay, because the redme I just read on the website describes how to connect a phone to my PC.... *ll be back on that in a second |
| 12:49.05 | WIMPy | mutax: The direction doesn't matter. |
| 12:49.23 | *** part/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 12:49.35 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 12:50.05 | WIMPy | However, if you don't need NT Mode you could also try capi. Or dahdi if your hardware is supported. |
| 12:50.44 | mutax | WIMPy: wait, you tell me this cheap USB device provides NT mode?! |
| 12:51.02 | WIMPy | mutax: Yes |
| 12:51.17 | mutax | c0000l |
| 12:52.00 | mutax | although I dont need it at the moment ;) |
| 12:52.10 | WIMPy | Just change interface.conf and use a crossover cable. |
| 12:54.32 | eppigy | CAN YOU FEEL THIS MAGIC IN THE AIR |
| 12:54.36 | WIMPy | I've tried two of them, on in TE and one in NT mode on my netbook. Works very well. |
| 12:54.40 | eppigy | IT MUST HAVE BEEN THE WAY YOU KISSED ME |
| 12:54.52 | plundra | Any suggestions on where to get help with Linksys SPA* provisioning? |
| 12:55.22 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:55.26 | redax | plundra: what kind of help? |
| 12:55.43 | mutax | [TK]D-Fender: http://pastebin.com/2X9WJNEF <- trace of a failed attempt |
| 12:55.52 | plundra | redax: When I resync a new profile, it doesn't clear the variables that are empty in the new one. The old value is left. |
| 12:56.21 | mutax | WIMPy: but first I have to get it working ;) |
| 12:56.37 | [TK]D-Fender | mutax: pastebin your complete CLI output, not wireshark 3rd party debug |
| 12:56.56 | mutax | [TK]D-Fender: can't provide that at the moment, sorry |
| 12:56.58 | *** join/#asterisk fcois93 (~Administr@bagnolet.acropolistelecom.net) |
| 12:56.59 | plundra | redax: If the new profile has <Profile_Rule_B></Profile_Rule_B>, and Profile_Rule_B is set, it will still stay set. Even when the rest of the values are from the new. |
| 12:57.04 | fcois93 | hello all |
| 12:57.07 | [TK]D-Fender | mutax: Because....? |
| 12:57.44 | mutax | [TK]D-Fender: the asterisk server is down and I really need to get it running with isdn first. |
| 12:57.46 | redax | plundra: uhh. never had such a problem ... have you tried with a space? |
| 12:57.51 | fcois93 | in the dialplan, I know that we can do XXX,n(priorityname)... but to going n+101, is it possible to do n(priorityname+101) ? |
| 12:58.04 | mutax | [TK]D-Fender: the trace is from the email i sent to sipgate |
| 12:58.13 | [TK]D-Fender | mutax: Down as in * is not running, or as in unaccessable at all? |
| 12:58.41 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:58.50 | [TK]D-Fender | fcois93: What is the point of that? |
| 12:58.58 | plundra | redax: I have not no, I'm basing me of the --xml-sample output and then changing the elements I want (python, lxml) |
| 12:59.14 | plundra | redax: I'll give space a try, although it feels horrible :-) |
| 12:59.21 | mutax | [TK]D-Fender: down as in not running :) yes, I could boot it up but I just dont have the time right now, sorry. |
| 12:59.54 | mutax | [TK]D-Fender: If you really want to debug this I can come back later when I have the response from sipgate and have time :) |
| 13:00.17 | [TK]D-Fender | mutax: You should bring up probelms when you're in a position to DO something about it. |
| 13:01.04 | [TK]D-Fender | mutax: Hint : their example is BAD |
| 13:01.10 | fcois93 | [TK]D-Fender: After a Dial and after the timeout (Dial(SIP/user/${§EXTEN},10)), I want to hangUp with the right message after a timeout (487), then, I need to know axeactly when a call pass the Dial line because of timeout |
| 13:01.11 | mutax | *argh* yes, true. but I am already working on the solution, only not right now. I thought It could be interesting to know that atm there sees no way to set the callerid. |
| 13:01.40 | *** join/#asterisk niros (~c075ec1d@gateway/web/freenode/x-ruuzgakiakpewebh) |
| 13:02.25 | Belgarath | [TK]D-Fender: http://pastebin.com/JTMU30E4 |
| 13:02.28 | mutax | [TK]D-Fender: why? |
| 13:02.29 | [TK]D-Fender | fcois93: then look at the ${DIALSTATUS} right after. I don't see a need for relative jumps from another label |
| 13:02.38 | Belgarath | it is output and configs on the botom |
| 13:02.48 | [TK]D-Fender | mutax: Read up a bit and come back when you're in a position to test |
| 13:02.53 | mutax | [TK]D-Fender: oky, now you made me bootup the asterisk server :) |
| 13:03.19 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 13:04.12 | [TK]D-Fender | Belgarath: Contact: <sip:0766091021@192.168.1.65> <--- you have not configured your server to work properly from behind NAT |
| 13:04.13 | [TK]D-Fender | ~sipnat |
| 13:04.14 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:04.16 | fcois93 | [TK]D-Fender: I want to be sure that the call pass Dial line because of the timeout |
| 13:04.21 | [TK]D-Fender | Belgarath: ^^^ go follow the guide |
| 13:04.30 | Belgarath | I did |
| 13:04.36 | [TK]D-Fender | fcois93: then look at the ${DIALSTATUS} right after. I don't see a need for relative jumps from another label <------------------ |
| 13:04.40 | [TK]D-Fender | Belgarath: You did it wrong |
| 13:04.42 | Belgarath | I have externip and nat=yes |
| 13:04.46 | Belgarath | probably |
| 13:04.56 | [TK]D-Fender | Belgarath: Otherwise I wouldn't be seeing your private IP in there |
| 13:04.59 | mutax | [TK]D-Fender: would you mind looking at my config, to be sure this is correct? I'll pastebin some part of my dialplan and the sipgate account setup... (as soon as the server is up) |
| 13:05.15 | [TK]D-Fender | mutax: Better |
| 13:05.35 | [TK]D-Fender | Belgarath: Also a 404 could be sent if your user is wrong, not just the target # dialed. |
| 13:06.04 | Belgarath | [TK]D-Fender: What do I need beside externip and nat=yes ? |
| 13:06.32 | [TK]D-Fender | Belgarath: Read. The. Guide. |
| 13:06.48 | redax | plundra: yeah. never tried the space stuff, just a wild though |
| 13:06.59 | [TK]D-Fender | Belgarath: And then when you think you've done everything, SHOW US a complete call along with yoru configs |
| 13:07.15 | fcois93 | [TK]D-Fender: the problem with the timeout is that after the timeout, Asterisk sendback a CANCEL! it is not RFC like |
| 13:07.32 | [TK]D-Fender | fcois93: * has concalled. Thats the way it is. |
| 13:07.33 | *** join/#asterisk coppice (~chatzilla@93.176.64.202.dyn.pacific.net.hk) |
| 13:07.39 | plundra | redax: A thought I have was throwing a comment in there, <!-- empty --> :) |
| 13:07.45 | [TK]D-Fender | fcois93: You don't get to choose how Dial cancels |
| 13:08.11 | [TK]D-Fender | cancelled* |
| 13:08.11 | fcois93 | [TK]D-Fender: in the RFC, after a dial timeout, Asterisk have to sendback a 487 |
| 13:08.26 | [TK]D-Fender | fcois93: Apparently not. |
| 13:08.37 | [TK]D-Fender | fcois93: *'s "comment" was "no thanks" apparently |
| 13:08.48 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:09.23 | fcois93 | [TK]D-Fender: then to be RFC like, I have to look at the DIALSTATUS and HangUp with the right code |
| 13:09.25 | *** join/#asterisk smooth_penguin (~smoove@59.95.16.111) |
| 13:09.26 | Belgarath | [TK]D-Fender: is there any reson why externip is not recognized by asterisk ? |
| 13:10.09 | Belgarath | because i have it in [general] section of sip.conf |
| 13:10.10 | [TK]D-Fender | fcois93: NO. You have NO choice in this. the cancel is sent inside of the dial. |
| 13:10.27 | [TK]D-Fender | Belgarath: Aren't learning, are you? |
| 13:10.31 | [TK]D-Fender | Belgarath: PASTEBIN |
| 13:10.50 | fcois93 | [TK]D-Fender: too bad. know you is it in the source? (which line in the Dial.c) |
| 13:11.06 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 13:11.06 | *** mode/#asterisk [+o file] by ChanServ |
| 13:11.08 | [TK]D-Fender | fcois93: Would be chan_sip.c |
| 13:11.13 | [TK]D-Fender | fcois93: Dial is agnostic |
| 13:11.34 | Belgarath | well i didn't want to apsetbin 3 lines but here you are : http://pastebin.com/iiXkF5Wb |
| 13:11.38 | plundra | redax: Hmm, do you remember from the top of your head, how to trigger a resync using another profile (b, c or d)? |
| 13:12.19 | [TK]D-Fender | Belgarath: thats ALL you have under [general]? |
| 13:12.24 | [TK]D-Fender | Belgarath: that doesn't look right... |
| 13:12.24 | Belgarath | no |
| 13:12.35 | [TK]D-Fender | Belgarath: yes, there IS more. You have clearly not followed the guide |
| 13:13.16 | redax | plundra: http://spa/admin/resync?[http|tftp]://URL/config.xml |
| 13:13.29 | redax | plundra: that's the forced one, what I remember actually |
| 13:13.56 | mutax | [TK]D-Fender: http://pastebin.com/NxSxRZER <= config parts) |
| 13:13.59 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 13:14.22 | mutax | [TK]D-Fender: right now in sip.conf I reset the callerid, so calls are possible |
| 13:14.34 | [TK]D-Fender | mutax: exten => _0X.,1,Set(CALLERID(all)="NUMBERTOSEND <NUMBERTOSEND>") <- quotes = illegal, and don't try setting the name |
| 13:14.37 | plundra | redax: Yeah, but I'm pretty sure you can make it use $Profile_Rule_B or something like that, oh well. Thanks anyway :-) |
| 13:15.16 | mutax | [TK]D-Fender: exten => _0X.,1,Set(CALLERID(num) = 021123456789) <- better? |
| 13:15.21 | Belgarath | [TK]D-Fender: http://pastebin.com/wNMEXKDh that's all |
| 13:15.26 | [TK]D-Fender | mutax: No.... |
| 13:15.29 | [TK]D-Fender | mutax: guess why |
| 13:15.50 | mutax | [TK]D-Fender: no idea |
| 13:16.01 | [TK]D-Fender | Belgarath: You didn't specify what is LOCAL. Go read the guide again |
| 13:16.08 | [TK]D-Fender | mutax: Spaces = ILLEGAL |
| 13:16.20 | [TK]D-Fender | mutax: exten => _0X.,1,Set(CALLERID(num) = 021123456789) <- BAD |
| 13:16.25 | [TK]D-Fender | mutax: exten => _0X.,1,Set(CALLERID(num)=021123456789) <- GOOD |
| 13:16.32 | mutax | [TK]D-Fender: whoa, omg |
| 13:16.53 | [TK]D-Fender | mutax: Functions & Set 101 |
| 13:17.06 | mutax | [TK]D-Fender: ? |
| 13:17.27 | [TK]D-Fender | mutax: Saying these are the basic global rules that were overlooked |
| 13:17.37 | Belgarath | [TK]D-Fender: whenever I specify localnet it stops working alltoghether |
| 13:17.50 | Belgarath | at least at the moemnt I have the incoming calls working |
| 13:18.16 | mutax | [TK]D-Fender: okay, now I remove fromuser in sip.conf but leave fromdomain to be sipgate.de? |
| 13:18.22 | Belgarath | also i have one iax provider that works ok |
| 13:18.24 | [TK]D-Fender | Belgarath: You're doing something wrong then. and I can't see if your peer to your ITSP is right, or the one to any of your phones is either. |
| 13:18.48 | Belgarath | i can make internal phone calls between the phones |
| 13:19.05 | Belgarath | and I could make outgoing calls before upgrade to 1.6 |
| 13:19.26 | [TK]D-Fender | Belgarath: You haven't configured the parts I do see properly, and there is plenty you aren't showing. |
| 13:19.37 | [TK]D-Fender | Belgarath: And there is no reason why this should work on any version. |
| 13:19.49 | [TK]D-Fender | Belgarath: Saying "it used to work" will not help you. |
| 13:19.52 | Belgarath | [Apr 13 14:19:17] NOTICE[2873]: chan_sip.c:22839 sip_poke_noanswer: Peer 'freespeech' is now UNREACHABLE! Last qualify: 33 |
| 13:20.08 | Belgarath | after adding localnet in general section of sip.conf |
| 13:20.09 | mutax | [TK]D-Fender: Received response: "Forbidden" ... as expected. '"Telefon 103" <sip:numbertosendwithoutspaces;)xxx@sipgate.de>;tag=as1df' |
| 13:20.13 | [TK]D-Fender | Belgarath: you are showing the error without showing the cause. |
| 13:20.21 | Belgarath | I do not know the cause |
| 13:20.30 | Belgarath | If I knew I wouldn't be here |
| 13:20.31 | [TK]D-Fender | bel the cause is what you SEND |
| 13:20.50 | [TK]D-Fender | Belgarath: The error is in response to packets you are sending, and I'm not seeing a pastebin anywhere. |
| 13:21.50 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 13:22.18 | Belgarath | http://pastebin.com/mzC1BdjS |
| 13:23.46 | Katty | morning |
| 13:25.51 | Belgarath | I only added localnet=10.0.0.0/8 to sip conf |
| 13:27.22 | plundra | redax: <var/> <-- No go, <var></var> clears it though. The fun/weird thing is, /admin/spacfg.xml uses <var/> tags :-) |
| 13:27.32 | plundra | If you were interested :-P |
| 13:27.37 | Belgarath | [TK]D-Fender: would you mind having a look ? |
| 13:27.39 | mutax | [TK]D-Fender: http://pastebin.com/gaHUFZjj <= trace from asterisk console, as requested |
| 13:28.34 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:28.58 | redax | plundra: hehh. good to know. thanks |
| 13:30.20 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-jbwgmmsatyyfwfin) |
| 13:34.42 | *** join/#asterisk supa_disko (~bleh@123-243-201-39.static.tpgi.com.au) |
| 13:34.44 | mutax | WIMPy: what version of lcr were you using? |
| 13:35.24 | smooth_penguin | hey Katty ! |
| 13:36.12 | WIMPy | mutax: git |
| 13:36.29 | supa_disko | for all US peoples, generally when people do a driving holiday |
| 13:36.44 | supa_disko | do they go east to west or the other way around? |
| 13:36.48 | mutax | WIMPy: I should have guessed that :) |
| 13:37.55 | WIMPy | But I think the 20090906 version was ok. |
| 13:38.09 | mutax | k |
| 13:40.56 | *** join/#asterisk jsjc (~chatzilla@115.131.211.38) |
| 13:41.44 | jsjc | I am trying to find some good examples of inbound call handleing, queues in dialplans. Anybody know any good source? |
| 13:42.26 | [TK]D-Fender | [09:22]<Belgarath>http://pastebin.com/mzC1BdjS <- this response is unimportant. It is only an options packet |
| 13:42.51 | Belgarath | [TK]D-Fender: would you have couple of minutes to help me ? |
| 13:42.59 | [TK]D-Fender | [09:27]<mutax>[TK]D-Fender: http://pastebin.com/gaHUFZjj <= trace from asterisk console, as requested <- now try again with enough VERBOSE so we can actually see what is being executed |
| 13:43.21 | Belgarath | so what output should I give you |
| 13:43.31 | Belgarath | that is the only thing that i got with debug enabled |
| 13:43.34 | Belgarath | when I put localhost |
| 13:43.41 | Belgarath | localnet (sorry) |
| 13:43.57 | Belgarath | and it says in sip show peers that peer is unreachable |
| 13:43.57 | mutax | [TK]D-Fender: I'll try. sip verbose or core verbose? |
| 13:44.00 | Belgarath | if I remove localnet |
| 13:44.07 | Belgarath | it is back to normal |
| 13:44.12 | [TK]D-Fender | Belgarath: I don't see your register attempt going out. I don't see a call attempt going out. I don't see a call trying to come in. |
| 13:44.16 | [TK]D-Fender | Belgarath: DO SOMETHING |
| 13:45.06 | *** join/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 13:45.07 | Belgarath | Well It was trying to register |
| 13:45.23 | Belgarath | wait |
| 13:45.47 | Belgarath | can i force it to register by hand ? |
| 13:45.52 | *** join/#asterisk smooth_penguin (~smoove@59.95.26.30) |
| 13:46.07 | mutax | [TK]D-Fender: I don't know what you expect to see in the verbose output... |
| 13:48.16 | [TK]D-Fender | Belgarath: reload SIP |
| 13:48.50 | [TK]D-Fender | mutax: dialplan applications being executed |
| 13:49.14 | mutax | okay, because then I dont need to copy paste that much. thx |
| 13:49.31 | Belgarath | http://pastebin.com/wntMXDek |
| 13:49.52 | Belgarath | [TK]D-Fender: I reloaded it multiple times and restarted asterisk twice by now |
| 13:50.56 | [TK]D-Fender | Belgarath: Reliably Transmitting (NAT) to 213.254.224.20:5060: <-- they AER NOT BEHIND nat. rEAD THE GUIDE AGAIN AND FIX THIS |
| 13:51.22 | Belgarath | they are from my perspective |
| 13:51.53 | Belgarath | if I am behind the nat and they are not I have to put nat=yes in the sip.conf corect ? |
| 13:51.57 | [TK]D-Fender | Belgarath: Your PROVIDER is not behind NAT. They never are. You always trust their IP's |
| 13:52.03 | mutax | [TK]D-Fender: http://pastebin.com/kzfbqpg6 thats all |
| 13:52.12 | [TK]D-Fender | Belgarath: Read. The. GUIDE. |
| 13:52.20 | Belgarath | I did |
| 13:52.27 | Belgarath | and it only added to the confusion |
| 13:52.38 | Belgarath | nice simple answer would be apriciated :) |
| 13:52.46 | [TK]D-Fender | Belgarath: Your PROVIDER is not behind NAT. They never are. You always trust their IP's <------------- |
| 13:52.48 | [TK]D-Fender | Belgarath: Your PROVIDER is not behind NAT. They never are. You always trust their IP's <------------- |
| 13:52.51 | mutax | Belgarath: no, incorrect then |
| 13:52.56 | [TK]D-Fender | Belgarath: ANSWER. Am I clear? |
| 13:53.09 | mutax | Belgarath: nat = no |
| 13:53.14 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:53.25 | Chainsaw | wb tzafrir |
| 13:53.43 | tzafrir_laptop | Chainsaw, hi |
| 13:53.58 | mutax | [TK]D-Fender: hmm. okay, now you know my mobile phone no ;o) |
| 13:54.09 | [TK]D-Fender | Belgarath: [a] <- in the guide. |
| 13:54.19 | Belgarath | i will personally kill the sip inventor as soon as i find him |
| 13:54.41 | mutax | Belgarath: *g* |
| 13:55.03 | mutax | Belgarath: throw a hardcopy of all related RFC's at him ;o) |
| 13:55.55 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:55.55 | Belgarath | mutax: IAX work without problem in the same setup why sip is so twisted that it cannot |
| 13:55.59 | Belgarath | :P |
| 13:56.37 | mutax | [TK]D-Fender: http://pastebin.com/aA5eMT6h latest one. wihtout my mobile no |
| 13:56.39 | mutax | :) |
| 13:56.53 | mutax | Belgarath: SIP was never intended to connect PBXs |
| 13:57.08 | mutax | Belgarath: to use sip phone at home behing nat it may work |
| 13:57.22 | mutax | Belgarath: NAT makes everything so complicated. |
| 13:57.23 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 13:57.37 | *** join/#asterisk mjsilva (~Manuel@mx1.connectinternational.eu) |
| 13:57.37 | *** join/#asterisk mdg (~mdg@unaffiliated/mgroman) |
| 13:57.43 | Belgarath | mutax: well other protocols can handle it pretty well |
| 13:57.51 | *** join/#asterisk jtexter3 (~jtexter3@72.242.229.213) |
| 13:58.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:58.06 | Belgarath | it is only twisted idea of having seperate data and control protocol that makes it so difficult |
| 13:58.15 | mutax | Belgarath: like in... ftp ;) |
| 13:58.16 | jtexter3 | Anyone ever see ast_playstream / ast_waitstream never return, but the called party doesn't hear the prompt? |
| 13:58.34 | mutax | jtexter3: you answered the line? |
| 13:58.36 | Belgarath | mutax: yes ftop is proime example |
| 13:58.41 | Belgarath | it didn;t work then welll |
| 13:58.50 | Belgarath | why diud somebody repeated their mistakes |
| 13:58.52 | jtexter3 | yes, outbound call, and I don't play until it's answered |
| 13:59.11 | mutax | Belgarath: from a programmer point of view its easier going woth two channels |
| 13:59.24 | Belgarath | I guess I will resign form my sip account and get another iax one |
| 13:59.24 | Belgarath | :P |
| 13:59.36 | mutax | Belgarath: and you can submit control data and payload at one ;) |
| 13:59.57 | mutax | jtexter3: the other party is behind nat? |
| 14:00.11 | mutax | jtexter3: in other words: does a normal call work? |
| 14:01.12 | mdg | Hi, I am trying to compile wctdm24xxp zaptel drivers from the 1.4.12 release, during compilation, it downloads some firmware for the driver, and it tries to install zaptel-fw-octal6114-064.bin, but cant find it so the installation fails. It looks like it is downloading zaptel-fw-vpmadt032.bin instead. |
| 14:01.28 | mdg | Do I need the firmware? Im unsure what it provides exactly |
| 14:01.47 | *** join/#asterisk rdircio (~admin@201.137.45.224) |
| 14:02.31 | Chainsaw | mdg: It programs the FPGA chips on the card. |
| 14:02.48 | Chainsaw | mdg: You will need that, because without it the card is dead in the water. It'll be like a car without a driver. |
| 14:03.21 | mdg | Chainsaw: ok thanks. Should I try to find the missing fw, or should I patch the makefile to use the file it is downloading ? |
| 14:04.09 | *** join/#asterisk moy (~chatzilla@74.12.133.16) |
| 14:07.54 | mutax | WIMPy: hmm. cant compile lcr atm because compat_af_isdn.h is missing and I don't know what provides this file. packages.debian.org doesn't know it either -.- |
| 14:08.25 | Chainsaw | mdg: Best to wget it in, yes. |
| 14:08.39 | mdg | Chainsaw: ok thanks again |
| 14:09.01 | Chainsaw | mdg: It should be in http://downloads.asterisk.org/pub/telephony/firmware/releases/ |
| 14:09.14 | jtexter3 | mutax, yeah, normal calls works |
| 14:09.41 | mdg | Chainsaw: got it |
| 14:09.47 | jtexter3 | what I don't understand is why ast_waitstream never returns. Even if the other party can't hear the prompt, ast_waitstream should return once it hits the end of the file |
| 14:09.53 | mutax | jtexter3: then my '1 week in asterisk' knowledge ends here. |
| 14:10.16 | mutax | jtexter3: tried setting core verbose? |
| 14:10.28 | mutax | jtexter3: this should tell you what happens. |
| 14:10.59 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:12.18 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:13.32 | *** part/#asterisk fcois93 (~Administr@bagnolet.acropolistelecom.net) |
| 14:14.11 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 14:16.19 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 14:17.07 | bent_screwdriver | Is the asterisk yum repo down? http://packages.asterisk.org/centos/5/current/i386/repodata/repomd.xml |
| 14:17.24 | mdg | Is the Asterisk bug tracker still at http://issues.asterisk.org/ ? |
| 14:18.20 | mdg | downforeveryoneorjustme.com says both sites are down |
| 14:18.23 | mdg | bent_screwdriver: ^ |
| 14:18.46 | *** join/#asterisk tiav (~tiav@mx.fr.smartjog.net) |
| 14:19.49 | bent_screwdriver | okay, this one works....baseurl=http://packages.digium.com/centos/$releasever/tested/$basearch/ |
| 14:22.12 | bent_screwdriver | mdg: but digium doesn't have much on their repo i guess.... |
| 14:22.19 | WIMPy | mutax: compat_af_isdn.h is in mISDNuser |
| 14:24.21 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 14:24.22 | leifmadsen | yes, stuff is down because a server has died on us |
| 14:24.57 | mdg | oh .. was going to file a bug report for that firmware makefile |
| 14:25.29 | leifmadsen | you won't be able to until someone goes onsite and is able to restart the server |
| 14:25.52 | leifmadsen | it's about 9:25am there, so someone should be on their way over now I'd think |
| 14:26.12 | WIMPy | leifmadsen: You keep your servers outside? |
| 14:26.23 | leifmadsen | Digium keeps their servers in a colo |
| 14:26.24 | mdg | No biggie, I got it fixed |
| 14:27.21 | mutax | WIMPy: no? at least not on the version I try to compile |
| 14:27.41 | mutax | WIMPy: which is a git co from it |
| 14:28.40 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-cktbqihtnqebelrx) |
| 14:29.36 | WIMPy | mutax: For me it went from /usr/src/mISDNuser/include/compat_af_isdn.h to /usr/include/mISDNuser/compat_af_isdn.h |
| 14:30.18 | mutax | WIMPy: it was missing in the checkout |
| 14:30.32 | mutax | WIMPy: lates tgz from the download section has it -.- |
| 14:30.53 | *** join/#asterisk kartik (~koolkarti@117.199.125.178) |
| 14:31.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:32.33 | WIMPy | mutax: Yes, I see it has vanished. It was still there when I fetched from git the last time. |
| 14:32.40 | *** join/#asterisk farkus (~chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:33.38 | mutax | WIMPy: yaaay - misdn-user and lcr compiled!!! misdn_log show traffic when calling me - now compile asterisk and then learn how to configre it all together;) WIMPy: you're great!!! |
| 14:33.50 | mutax | WIMPy: thx for your patience with me |
| 14:34.24 | WIMPy | mutax: That's not patience. I've been at Aldi inbetween :-) |
| 14:34.29 | kazaa_lite | hi a;; |
| 14:34.33 | mutax | *lol* |
| 14:35.14 | WIMPy | mutax: You probably missed chan_lcr wehn you installed lcr before asterisk. But you can try with lcr on it's own already. |
| 14:35.41 | mutax | i am in the process of recmpilingasterisk right now... |
| 14:35.52 | kazaa_lite | how can i save audio in 16Khz, while my g722 phone uses 8Khz:( |
| 14:36.22 | *** join/#asterisk smooth_penguin (~smoove@59.95.28.135) |
| 14:38.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:38.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:38.35 | coppice | that's not a phone. its a phoney |
| 14:40.19 | *** join/#asterisk nickfennell (~nick@195.137.23.30) |
| 14:41.51 | kazaa_lite | its snom300 |
| 14:42.22 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 14:43.23 | coppice | they are all phoneys. the RTP spec demands that they be. they are all 16kHz really |
| 14:44.02 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 14:44.04 | bent_screwdriver | has anyone here ever submitted a feature request to Polycom? |
| 14:44.23 | raden_work | is there a way to playback all wavs in a directory ? |
| 14:45.04 | kazaa_lite | coppice: what do you mean by phoneys?:( |
| 14:45.31 | coppice | they say 8k in the SDP, but are really 16k |
| 14:45.49 | mutax | raden_work: linux commandline or asterisk? |
| 14:45.59 | raden_work | asterish |
| 14:46.18 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 14:46.58 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 14:47.39 | kazaa_lite | coppice: ok... here is what i want to see 8k and 16k g722 working and what different it causes.... how can i test this thing.... if smon300 is actually 16k, then how can i test 8k thing? |
| 14:47.48 | WIMPy | kazaa_lite: I don't know the whole story, but 16KHz needs to be declared as 8KHz in order not to break thigs. |
| 14:48.49 | coppice | kazaa_lite: there is no such thing as 8k G.722. it is always 16k. its just tagged as 8k on the SDP |
| 14:49.07 | WIMPy | Isn't G.722 always 16KHz? (or more precisely 16Ks/s |
| 14:49.24 | mutax | WIMPy: lcr: symbol lookup error: lcr: undefined symbol: init_af_isdn *grrrrrr* |
| 14:49.31 | kazaa_lite | then why does asterisk g722 code has two sections: one for g722 and slin8 and one for slin16? |
| 14:49.33 | mutax | WIMPy: this is drivin me crazy... |
| 14:49.46 | *** join/#asterisk smooth_penguin (~smoove@59.95.28.135) |
| 14:49.48 | WIMPy | mutax: Where does that happen? |
| 14:49.52 | coppice | slin8 means 8 bit, not 8k |
| 14:50.06 | mutax | lcr start. I think I know why... |
| 14:50.25 | raden_work | mutax, any idea ? |
| 14:50.25 | mutax | WIMPy: let me investigate. I thin an old so file: /usr/local/lib/libmisdn.so |
| 14:50.40 | kazaa_lite | ahhhhhh..... right....... thanks for explaining:) |
| 14:51.07 | kazaa_lite | and how can i see the path for g722 and 16bit in use? |
| 14:51.30 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:52.51 | mutax | WIMPy: lcr is running *phew* ;o) Old libs in /lib64 -.- my fault |
| 14:53.18 | massoud | Hey there, |
| 14:53.32 | massoud | do you guys know a svn mirror for asterisk-dev server ? |
| 14:53.51 | massoud | it seems that the official digium svn server is down currently |
| 14:54.00 | WIMPy | mutax: Time for a 'lcradmin state' and some nice colours ;) |
| 14:54.14 | Faustov | massoud: you might be interested in #strowger |
| 14:54.18 | *** join/#asterisk dwayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:54.34 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 14:54.38 | massoud | Faustov: what is it ? |
| 14:55.02 | Faustov | http://www.strowger.co.za |
| 14:55.11 | Dovid | anyone from Turkey here ? |
| 14:55.58 | massoud | Faustov: and there is a svn somehwere ? |
| 14:56.50 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 14:57.17 | Faustov | massoud: it has a git + you might be interested because it is an asterisk fork with many overdue patches |
| 14:58.22 | massoud | Faustov: yes :), do you have the git url for cloning ? |
| 14:58.42 | Faustov | <PROTECTED> |
| 14:58.48 | manxpower | what a delightful placeholder page |
| 14:59.09 | Faustov | oh, still not up |
| 14:59.14 | Faustov | well it's a new thing |
| 14:59.16 | Faustov | sorry |
| 15:00.34 | kazaa_lite | coppice: any suggestion? |
| 15:00.40 | Dovid | hello ev1 |
| 15:00.43 | Dovid | morning TK |
| 15:00.46 | WIMPy | Faustov: So how do we find out about the differences? |
| 15:01.15 | raden_work | is there a way to playback all sound files in a directory by using a wildcard or something ? |
| 15:01.48 | Faustov | WIMPy: by using the bugzilla: http://bugs.strowger.co.za/ |
| 15:01.59 | Faustov | btw, it's not mine |
| 15:02.17 | WIMPy | That was a better url |
| 15:02.42 | [TK]D-Fender | Belgarath: You still have not shown me proper configs and debug |
| 15:02.49 | [TK]D-Fender | Belgarath: it works fine for everyone else |
| 15:05.12 | raden_work | [TK]D-Fender, any idea how to playback all sound files in a directory ? |
| 15:05.33 | raden_work | like wildcard ? /var/lib/asterisk/sound/*.wav ? |
| 15:05.59 | kazaa_lite | which audio format is 16bit(slin16)? |
| 15:06.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:06.14 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:08.06 | *** join/#asterisk garymc (~chatzilla@host81-148-11-241.in-addr.btopenworld.com) |
| 15:09.46 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.66) |
| 15:13.15 | redax | is there anybody using mISDN with recent kernels and asterisk? |
| 15:13.25 | Belgarath | [TK]D-Fender: I guess I am very unlucky :) |
| 15:13.27 | redax | either using chan_misdn or chan_lcr ? |
| 15:13.29 | mutax | redax: I am in the procsess of setting this up |
| 15:13.36 | mutax | redax: chan_lcr |
| 15:13.42 | Belgarath | but I created speparate entry for incoming and outgoing connections |
| 15:13.43 | WIMPy | redax: Yes |
| 15:13.45 | Belgarath | and it works now |
| 15:13.53 | Belgarath | without localnet bit |
| 15:14.18 | mutax | WIMPy: something hangs up here :( lcr gets the call, forwrds it to asterisk but something is broken |
| 15:14.21 | *** join/#asterisk edwin_quijada (~macaruchi@200.26.172.50) |
| 15:14.25 | edwin_quijada | Hi! |
| 15:14.38 | mutax | WIMPy: I cant here audio, the call seems to be hangup immedeately |
| 15:14.45 | TJ^ | PCI-X |
| 15:14.50 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 15:14.55 | TJ^ | thats compatible with a normal PCI card right? |
| 15:15.05 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:15.22 | WIMPy | mutax: Log? |
| 15:15.51 | redax | aaa. unfortunetly I _HAVE_ to use kernel 2.6.32, and the latest mISDN (downloaded from linux-call-router) not compiling |
| 15:15.53 | WIMPy | mutax: Or: Does it get to the right extension at all? |
| 15:16.20 | redax | is it ok, to download mISDN from misdn.org and compile lcr with that? |
| 15:16.31 | mutax | WIMPy: http://pastebin.com/ApU0xRNX |
| 15:16.41 | WIMPy | redax: With recent kernels do NOT download mISDN, only mISDNuser. |
| 15:16.51 | mutax | redax: use the tgz not the git checkout |
| 15:16.58 | mutax | redax: I a using 2.6.33 |
| 15:17.19 | WIMPy | redax: And we just found out that lcr won't compile with the latest git version of mISDNuser. |
| 15:17.19 | mutax | WIMPy: it goes to the right extension, config follows |
| 15:17.23 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 15:17.23 | redax | mutex: nah. so shall I download mISDN tarball as well ? |
| 15:17.29 | redax | sorry mutax, |
| 15:18.56 | mutax | WIMPy: http://pastebin.com/t7tga5nU |
| 15:19.23 | mutax | redax: you only need lcr and misdn-user |
| 15:19.32 | mutax | redax: misdn is part of the kernel |
| 15:19.46 | mutax | redax: its in device-drivers, ISDN somewhere |
| 15:20.24 | WIMPy | mutax: Is mISDN_dsp loaded? |
| 15:20.26 | mutax | CONFIG_MISDN=m .... |
| 15:20.32 | *** join/#asterisk Fubard (~brawr@173.10.235.205) |
| 15:20.45 | redax | ok. my kernel has mISDN modules |
| 15:20.47 | mutax | yes |
| 15:20.55 | mutax | WIMPy: Ill reloadit |
| 15:21.04 | redax | compiled mISDNuser. what to do, no mISDN and misdn-init at all ... |
| 15:21.27 | edwin_quijada | I have a weird problem using callfiles when I send to outgoing a callfile this happens as it works but dont. This is my callfile and extension http://pastebin.com/zVU1XbAa |
| 15:21.36 | WIMPy | mutax: Hmm. I don't see a reason. Is there anything else in the lcr log perhaps? |
| 15:21.36 | mutax | redax: execute misdn_misdn_info |
| 15:22.00 | mutax | WIMPy: ill pastebin lcr log in a second. j |
| 15:22.38 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 15:23.00 | edwin_quijada | I am trying to do a call with callfile but it doesnt work.... |
| 15:23.02 | edwin_quijada | http://pastebin.com/zVU1XbAa |
| 15:23.04 | redax | cannot open mISDN due to Address family not supported by protocol. ... |
| 15:23.18 | mutax | WIMPy: nargh. my wife got home and is blocking the phoneline so I cant do testcalls right now ;o) |
| 15:23.42 | mutax | redax: aore you sure you have ISDN enabled in your kernel? |
| 15:23.54 | mutax | zcat /proc/config.gz | grep -i ISDN |
| 15:23.59 | mutax | if you have config-support |
| 15:25.05 | WIMPy | redax: You might want to make sure, you don't have any old misdn stuff around. I think it didn't do any harm for me, but it's incompatible. |
| 15:25.07 | redax | CONFIG_ISDN=y, CONFIG_MISDN=m CONFIG_MISDN_DSP=m, CONFIG_MISDN_L1OIP=m |
| 15:25.07 | redax | ... |
| 15:25.42 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 15:25.46 | *** join/#asterisk RobH (~robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 15:26.06 | mutax | WIMPy: http://pastebin.com/wQYqa1fs |
| 15:26.14 | WIMPy | redax: Do you have the modules loaded? What gave the error? |
| 15:26.19 | edwin_quijada | asterisk tell me that it did the call uisng callfile but it |
| 15:26.20 | mutax | redax: I had old misdn.so files in /usr/lib and /lib |
| 15:26.33 | redax | ok. loaded mISDN_core, and mISDN_dsp ok |
| 15:27.09 | WIMPy | redax: Some device might help. |
| 15:27.12 | redax | loaded hfcmulti |
| 15:27.24 | redax | now I have all the ports.. :D |
| 15:28.30 | *** join/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 15:28.51 | redax | where does this new mISDN subsys stores the configuration for ports? |
| 15:29.15 | mutax | WIMPy: I'll be back in a minute... |
| 15:29.17 | WIMPy | redax: It's in lcr / interface.conf |
| 15:29.53 | WIMPy | mutax: You're on a ptp or ptmp line? I see overlap messages there. |
| 15:30.24 | redax | ptp and NT |
| 15:31.50 | raden_work | is there a way to have like a line indicator on my phone to tell if another extension is in use or something aastra 9133i |
| 15:32.51 | mutax | WIMPy: I am on this analog to isdn adapter-"line" -.- |
| 15:32.53 | redax | WIMPy: where should I put the lcr/default files? |
| 15:33.49 | mutax | WIMPy: ptmp I guess? |
| 15:33.53 | WIMPy | mutax: Oh, I remember. But that must be ptmp I'd guess? |
| 15:34.19 | WIMPy | redax: By default /usr/local/lcr |
| 15:34.20 | mutax | yes, in the manual they say one can attach multiple phones |
| 15:34.25 | *** part/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 15:34.35 | mutax | WIMPy: what does the overlap mean? |
| 15:35.07 | WIMPy | mutax: That meant lcr is requesting mor digits, i.e. it expects DDI. |
| 15:35.10 | redax | yeah /usr/local/lcr |
| 15:35.25 | mutax | hmm. okay. I'll see the lcr config |
| 15:35.41 | WIMPy | mutax: So you need to make sure the number in the setup message matches an extension in your dialplan. |
| 15:35.52 | mutax | in lcr or asterisk? |
| 15:36.12 | mutax | in lcr i set default to asterisk, in asterisk i have the two internal MSNs as 1 and 2 |
| 15:36.12 | WIMPy | mutax: Your choice. |
| 15:36.36 | mutax | the adaptor emits calls to these msn's for line 1 and 2 |
| 15:36.36 | WIMPy | mutax: I'd only configure 'remote=asterisk' in lcr and do the rest in Asterisk. |
| 15:36.52 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:37.02 | mutax | you can configure remote=asterisk and in additon a context, right? |
| 15:37.14 | WIMPy | correct |
| 15:37.19 | redax | WIMPy: after `lcr start' ? |
| 15:37.36 | WIMPy | mutax: You need your complete phone number as received as an extension. |
| 15:37.42 | [TK]D-Fender | edwin_quijada: what card are you using? |
| 15:38.08 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 15:38.10 | mutax | WIMPy: this woul be '1' in this case |
| 15:38.28 | mutax | WIMPy: but I dont think thats the problem... see my extension.conf... |
| 15:38.34 | WIMPy | mutax: In you log you have 03xxxx8 |
| 15:38.41 | mutax | http://pastebin.com/t7tga5nU |
| 15:38.43 | WIMPy | s/you/your/ |
| 15:38.50 | mutax | WIMPy: thats the callerid of the incoming call |
| 15:39.00 | WIMPy | redax: What after that? |
| 15:39.27 | mutax | WIMPy: and this analog to isdn adapter emits a call for msn '1' or '2' depending on the analog line the call comes in |
| 15:39.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:39.43 | WIMPy | Right. Traces are easier to read than logs... |
| 15:40.24 | redax | WIMPy: configured interface.conf... had a [PBX] entry, and addedd portnum[0-7] with nt |
| 15:40.35 | mutax | WIMPy: and i asterisk I have 4 rules. exten=>_s,1,Goto(extern_context,s,1) then s and 1 and 2 - so all should get routed together |
| 15:40.40 | redax | WIMPy: so shall I start `lcr start' ? |
| 15:41.46 | WIMPy | mutax: I think you should check yur interface config. Overlap receive shouldn't happen in your setup. |
| 15:42.10 | WIMPy | redax: What are you waiting for? |
| 15:42.51 | mutax | WIMPy: thats my interface.conf: [Ext] \n portnum 0 |
| 15:43.14 | Naikrovek | raden_work: that's easy on polycom :D |
| 15:43.36 | Naikrovek | raden_work: i think what you're looking for is BLF, but Aastra may call it something different |
| 15:43.44 | redax | WIMPy: loading chan_lcr.so in asterisk drops error |
| 15:44.19 | raden_work | Naikrovek, got it |
| 15:44.41 | [TK]D-Fender | raden_work: IIRC you can allocate one of the 4 call appearances on the 9133 to a speed-dial+BLF |
| 15:44.46 | WIMPy | mutax: I don't have more there, either, but am wondering now if a 'ptmp' or such might make sense. |
| 15:45.14 | WIMPy | redax: Try 'lcaramin state' |
| 15:46.39 | redax | whoa. now it's loaded |
| 15:49.31 | redax | WIMPy: does chan_lcr has some kind of configfile at /etc/asterisk ? |
| 15:49.58 | WIMPy | redax: no |
| 15:50.27 | *** join/#asterisk Z_God (~julius@wlan230008.mobiel.utwente.nl) |
| 15:51.34 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 15:52.53 | redax | WIMPy: then what context shall the incoming calls go? |
| 15:53.01 | kazaa_lite | how to enable L16 PCM Wideband codec in asterisk? |
| 15:53.15 | kazaa_lite | i am using xlite |
| 15:53.29 | WIMPy | redax: Where you want them. |
| 15:53.30 | Qwell | is that just slin16? |
| 15:54.19 | kazaa_lite | Qwell: i dont know... but I need some slin16 audio format on some phone and this name seemed a close match among the available codecs in the phone |
| 15:54.31 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 15:54.49 | redax | WIMPy: [lcr-incoming] or whatever.. , just asking where can I specify the context... |
| 15:54.50 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:55.14 | WIMPy | redax: in lrcs routing.conf |
| 15:55.51 | idespinner | kazaa_lite, just do a allow=all in the peers for your 2 phones, asterisk will allow passthrough if both xlite and your mystery phone support it |
| 15:56.12 | mutax | WIMPy: 004176 (in set_tone() line 1465): isdn port now plays tone:'dialing'. <- wth is it playing tones, I disabled them in the vconfig |
| 15:56.55 | manxpower | you almost never want allow=all |
| 15:57.01 | *** join/#asterisk cjk (~cjk@80.90.44.29) |
| 15:57.12 | manxpower | Asterisk will happily allow codecs it cannot transcode |
| 15:57.29 | WIMPy | mutax: Yes, something strange is going on there. But I'm not sure, what. Have you tried making an outgoing call, yet? |
| 15:58.41 | mutax | WIMPy: no, because this is my notebook acting as asterisk. my original asterisk 'server' would need a new kernel for that and I think USB1 is too slow for isdn? dunno. However, I'll try that later I think.... strange... |
| 15:58.59 | jsjc | hehe started reading on queues! this is a new world!! woooww.... nice! ;) |
| 15:59.42 | mutax | WIMPy: asterisk seems to sendthe overlap message |
| 15:59.45 | WIMPy | mutax: Just all yourself an play the current time or something. And USB 1 should be enough for 144kbit/s. |
| 16:00.00 | WIMPy | call |
| 16:00.09 | mutax | WIMPy: ahh, using a callfile. yes, that could work!!! ;) |
| 16:00.19 | mutax | bangs his head on the table |
| 16:00.38 | WIMPy | mutax: Or just 'channel originate' from the cli. |
| 16:00.55 | redax | WIMPy: how to setup a ruleset to route every call to asterisk ? tried `default : remote application=asterisk |
| 16:02.34 | WIMPy | redax: That should be it. I can't remember why, but I've got 'remote=asterisk : remote application=asterisk' there. |
| 16:04.39 | redax | hm. I don't understand this intern/extern stuff in routing.conf |
| 16:05.35 | redax | what if I don't want extensions at all.. just bypass the calls coming from asterisk to the isdn interfaces, and vica versa |
| 16:06.07 | WIMPy | redax: Then you don't need it. Just put that line under [main]. |
| 16:07.34 | WIMPy | redax: You wanted to use your interfaces in NT mode? |
| 16:08.10 | redax | es |
| 16:08.11 | redax | yes |
| 16:09.10 | WIMPy | redax: Then you probably want to be able to select the ports. |
| 16:09.41 | WIMPy | For that you need to configure them seperately in both interface.conf and routing.conf. |
| 16:10.32 | mutax | WIMPy: channel originate LCR/0/xxxxx extension 1234 should be correct? |
| 16:11.08 | mutax | WIMPy: 006234 trace 13.04.10 18:10:08.476 EP(2): INTERFACE (too busy) port 0 interface 0 |
| 16:11.22 | WIMPy | mutax: Depends on your extension 1234, but might be ok. |
| 16:11.26 | mutax | WIMPy: sth. here is borken |
| 16:11.41 | mutax | WIMPy: ots the default asterisk setup woth a demo user/mailbox@12344 |
| 16:11.57 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:11.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:12.17 | WIMPy | mutax: What's being signalled? |
| 16:12.33 | mutax | where? |
| 16:12.51 | mutax | at the console? nothing |
| 16:13.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:13.04 | raden_work | there a list of BLF notifys somewhere ? |
| 16:13.36 | WIMPy | mutax: is 0 the correct name of your port? |
| 16:13.51 | mutax | WIMPy: how do I find out? |
| 16:14.08 | mutax | WIMPy: 000000 trace 13.04.10 18:12:09.765 CH: PORT (open) port 0 mode terminal channels 2 |
| 16:14.12 | WIMPy | It's the section name in interface.conf |
| 16:14.17 | mutax | ahhhhhh!!!! |
| 16:15.06 | mutax | heh! |
| 16:16.09 | mutax | i am getting complete audio garbage there |
| 16:16.14 | mutax | then it hangs up |
| 16:17.24 | WIMPy | Now that's getting really interesting :-( |
| 16:18.31 | mutax | hmm. i cant repeat it. now it refuses to pick up the line and dial out.... |
| 16:19.48 | WIMPy | cause code? |
| 16:20.07 | WIMPy | Anyway: That really sounds borked :-( |
| 16:20.39 | mutax | cause_42 |
| 16:20.44 | mutax | what else -.- |
| 16:20.53 | WIMPy | That's decimal? |
| 16:21.07 | mutax | [Apr 13 18:20:26] WARNING[12561]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'Console' (cause 66 - Channel not implemented) |
| 16:21.24 | mutax | okay. I think there are many fuckups here |
| 16:21.53 | mutax | one is the audio codec the other the not existing extn 1234 the other the lockup of the analoge isdn adapter |
| 16:21.55 | WIMPy | 'channel TYPE'??? It's getting more and more interesting. |
| 16:22.58 | manxpower | Console is the sound card in the server. |
| 16:23.05 | redax | WIMPy: is there a way to configure the signalling? like etsi/euroisdn and similar? |
| 16:23.17 | WIMPy | But... As you had audio problems, maybe it's back to the simple things. Maybe you should try another cable. |
| 16:23.50 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.233.91.dsl.dyn.forthnet.gr) |
| 16:24.22 | WIMPy | redax: That would be in interface.conf, but I'm not sure it has much to choose from. |
| 16:25.11 | redax | lcr interface really don't show anything regarding signalling |
| 16:28.13 | kazaa_lite | if i dont attend call why do i get: Received SIP subscribe for peer without mailbox: 1006 |
| 16:28.23 | kazaa_lite | or how to setup mailbox for 1006? |
| 16:28.37 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 16:28.54 | pabelanger | OT: anybody else having gmail.com problems? |
| 16:30.12 | WIMPy | redax: Looks like you actually have no choice, except for that 'te-special'. |
| 16:30.49 | jozu | please, i have a dude |
| 16:30.56 | jozu | how to insert callerid from my DID? |
| 16:31.03 | jozu | i have only this |
| 16:31.06 | jozu | exten => 3000,1,Dial(SIP/siptosisuser) |
| 16:31.16 | raden_work | anyone know how to get the BLF on a aastra phone to work when another extension is on the phone ? |
| 16:31.43 | p3nguin | jozu: That shows that you are dialing OUT. DID is for dialing IN. |
| 16:31.57 | bent_screwdriver | exten => 3000,1,Set(CALLERID(num)=5555555555) |
| 16:32.25 | jozu | mmm, this is for dialing out... |
| 16:32.34 | p3nguin | and DIDs are for dialing in. |
| 16:32.38 | jozu | yes |
| 16:32.43 | p3nguin | ~did |
| 16:32.44 | infobot | did is, like, Direct Inward Dialing, or just a phone number |
| 16:33.08 | mutax | WIMPy: I think this is the perfect time for a reboot: |
| 16:33.08 | mutax | Module Size Used by |
| 16:33.08 | mutax | hfcsusb 23202 4294967275 |
| 16:33.09 | raden_work | only thing I can get to work is to show the BLF indicator when a line is on hold :( |
| 16:33.29 | mutax | WIMPy: at least the audio codec is broken. |
| 16:33.31 | mutax | brb |
| 16:33.57 | WIMPy | mutax: I also have that problme, but it doesn't do any harm for me. |
| 16:34.05 | jozu | mmm... |
| 16:35.04 | jozu | i have DID with [xxx] context and the first line in the [xxx] context is exten => 3000,1,Dial(SIP/siptosisuser) |
| 16:35.53 | p3nguin | okay |
| 16:36.08 | jozu | oks? i want to see who is calling my DID (p-asserted-identity?) |
| 16:36.10 | manxpower | jozu: What is your specific issue? |
| 16:36.31 | p3nguin | Incoming caller id doesn't work? |
| 16:36.32 | *** join/#asterisk mmoreno80 (~matt@190.190.47.146) |
| 16:36.35 | mmoreno80 | hi. |
| 16:36.35 | manxpower | jozu: Unless you disable it, Caller*ID is AUTOMATICALLY PASSED. |
| 16:36.36 | jozu | no |
| 16:36.46 | manxpower | Remember, many carriers do not even provide callerid service on voip numbers. |
| 16:37.09 | jozu | yes, but i see the number in the sip debug |
| 16:37.29 | manxpower | jozu: pastebin the CLI output with sip debug of a problem call. someone may be able to help you |
| 16:37.31 | WIMPy | Hmm. I managed to unload hfcsusb howver. There must be some difference somewhere. |
| 16:37.38 | mmoreno80 | can * play wav files? I'm trying to convert wav to gsm, but result isn't good at all. |
| 16:37.47 | jozu | thanks :) |
| 16:37.51 | manxpower | mmoreno80: yes! |
| 16:38.09 | WIMPy | mmoreno80: Yes, unless they contain metadata. |
| 16:38.11 | manxpower | but they must be mono, 8khz, 16bit, IIRC |
| 16:38.56 | mmoreno80 | manxpower: there is a way to convert a wav to another wav with all that requirements? |
| 16:39.12 | mmoreno80 | manxpower: btw, I'm using sox to convert. |
| 16:39.27 | manxpower | mmoreno80: yes, thousands of people use sox to do it every day. Perhaps you could look at the Wiki for (maybe correct) examples |
| 16:39.53 | bmoraca_work | how often does an asterisk box look at SRV records to verify that their current target box is up? |
| 16:40.04 | mmoreno80 | manxpower: ok. can you provide me with wiki url? |
| 16:40.26 | manxpower | ~answers |
| 16:40.27 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 16:40.39 | mutax | WIMPy: hah! the kernel was f. up |
| 16:40.41 | p3nguin | I prefer to use asterisk's internal convertor. |
| 16:40.43 | mmoreno80 | manxpower: thanks! |
| 16:40.50 | mutax | WIMPy: lcradmin can make a testcall |
| 16:40.57 | mutax | WIMPy: after reboot it works |
| 16:40.58 | manxpower | p3nguin: that is only on 1.6.x and later |
| 16:41.04 | mutax | now I can go back to asterisk ;) |
| 16:41.09 | p3nguin | file convert infile.wav outfile.gsm |
| 16:41.13 | p3nguin | and I do this on 1.4, since that's what I use. |
| 16:41.14 | WIMPy | mutax: cool |
| 16:41.39 | mutax | WIMPy: at least if I find new batteries for my DECT phone *argh* too many testcalls -> empty |
| 16:41.47 | manxpower | p3nguin: I sit corrected. |
| 16:42.37 | *** part/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 16:43.45 | mutax | WIMPy: hmmm. testcall was okay. now its broken again..... |
| 16:44.54 | mutax | Module Size Used by |
| 16:44.54 | mutax | hfcsusb 23202 4294967294 |
| 16:45.02 | mutax | the kernel driver is broken I'd say |
| 16:45.19 | WIMPy | Yes. I have similar counts. |
| 16:45.31 | mutax | wtf? |
| 16:45.47 | mutax | this is a major bug... |
| 16:46.07 | WIMPy | nfi. Something counts wrong there as it looks like. |
| 16:46.38 | WIMPy | The interesting thing is that I have been able to unload the driver on my netbook. So there must be a difference somewhere. |
| 16:46.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:47.16 | mutax | is it a 32bit machine? |
| 16:47.17 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 16:47.45 | mutax | now i was able to unload too |
| 16:47.50 | WIMPy | Both |
| 16:48.12 | WIMPy | Both machines that is |
| 16:48.20 | mutax | how do you load the modules |
| 16:48.34 | mutax | I mean what parameters do you give to modprobe |
| 16:48.50 | WIMPy | none |
| 16:49.14 | mutax | hmm. okay. you can set e.g. the audio codec on misdn_dsp |
| 16:51.05 | raden_work | there anything special i have to use to trigger a BLF with asterisk when a phone is in use ???? |
| 16:51.46 | manxpower | raden_work: of course there is! |
| 16:51.54 | raden_work | whats that ? |
| 16:52.01 | manxpower | raden_work: perhaps you missed the page on BLF in the Wiki |
| 16:52.05 | WIMPy | raden_work: You have to allow subscriptions and provide hints in the right context. |
| 16:52.12 | raden_work | got that |
| 16:52.25 | raden_work | allowsubscribe=yes |
| 16:52.29 | p3nguin | Then you have to actually subscribe to those hints. |
| 16:52.45 | raden_work | p3nguin, howso ? |
| 16:53.05 | raden_work | subscribecontext=internal |
| 16:53.20 | p3nguin | The phone has to subscribe. |
| 16:53.32 | manxpower | raden_work: perhaps you missed hints and the buddy watch on your Polycom phones |
| 16:53.44 | manxpower | raden_work: you need to read the WHOLE page, not just the first paragraph! |
| 16:53.45 | raden_work | aastra phones |
| 16:53.46 | Naikrovek | he has aastra phones |
| 16:53.51 | raden_work | 9133i |
| 16:54.03 | raden_work | only get a indicator lamp when someone is put on hold on that extension thats it |
| 16:54.08 | Naikrovek | another reason to use polycom : the support in this channel for those phones is outstanding |
| 16:54.12 | manxpower | Naikrovek: I should have known. He never reads any docs and seems to be constantly asking stupid questions. |
| 16:54.13 | raden_work | exten => 101,hint,SIP/101 |
| 16:54.38 | Naikrovek | manxpower: he's very clever, except with the aastra thing |
| 16:54.39 | Naikrovek | :P |
| 16:54.39 | raden_work | manxpower, I read plenty |
| 16:54.43 | raden_work | followed examples |
| 16:54.45 | raden_work | esearched |
| 16:54.46 | p3nguin | So now a subscription for exten 101 would check SIP/101's state. |
| 16:54.47 | raden_work | and still |
| 16:55.19 | *** join/#asterisk Weazel (~something@net.kolcore.com) |
| 16:55.27 | raden_work | p3nguin, is there something special i have todo to subscribe ? |
| 16:55.36 | p3nguin | Yes. |
| 16:55.37 | raden_work | ? |
| 16:55.39 | p3nguin | Subscribe. |
| 16:55.48 | Naikrovek | that's phone dependent though |
| 16:55.50 | Naikrovek | the subscritpion |
| 16:55.56 | Naikrovek | subscription* |
| 16:56.10 | raden_work | why do i get a indicator when someone put on hold ???? |
| 16:56.14 | raden_work | and no other time |
| 16:56.52 | manxpower | raden_work: What exactly have you done to the Aastra configuration to make it subscribe to the hints |
| 16:57.30 | raden_work | what the aastra manual said set the drop down box to BLF and put extension in value |
| 16:57.34 | raden_work | book says no more than that |
| 16:57.42 | raden_work | looked up a howto had same thing nothing more |
| 16:58.13 | manxpower | so you put in 101 ? |
| 16:58.41 | raden_work | yes |
| 16:58.57 | raden_work | and only time light will come on is if 101 puts a call on hold |
| 16:58.59 | manxpower | Maybe it wants a URI instead? |
| 16:59.09 | p3nguin | 101@internal ? |
| 16:59.13 | raden_work | ill try that |
| 16:59.28 | manxpower | raden_work: I always needed things like limitonpeer and stuff like that. |
| 16:59.35 | p3nguin | That's how Asterisk displays hints if you run "core show hints". |
| 16:59.43 | WIMPy | To me that sounds like the subscription is working, but the phone is just not interested in other states. |
| 16:59.45 | raden_work | manxpower, thats set |
| 17:00.02 | raden_work | WIMPy, i agree |
| 17:00.36 | manxpower | raden_work: maybe you need to post your sip.conf and hope someone can help you |
| 17:00.46 | WIMPy | raden_work: Do you see the extensions ringing? |
| 17:00.58 | manxpower | core show hints should show all the state information, even if the phone does not |
| 17:00.59 | raden_work | WIMPy, NO I DONT |
| 17:01.26 | manxpower | WIMPy: see it ringing where? CLI? the phone? the hints? |
| 17:01.37 | raden_work | the phone BLF indicator |
| 17:01.49 | manxpower | raden_work: try "core show hints" |
| 17:01.50 | WIMPy | raden_work: Maybe your phone has settings to tell it what states to signal in what way, like steady light/fast/slow blinking? |
| 17:02.24 | raden_work | <PROTECTED> |
| 17:02.29 | raden_work | 101 is on a call right now :( |
| 17:02.33 | *** part/#asterisk Weazel (~something@net.kolcore.com) |
| 17:02.36 | manxpower | raden_work: then your asterisk is not configured correctly. |
| 17:02.48 | p3nguin | bingo |
| 17:02.49 | manxpower | sounds like lack of your version of limitonpeer (not used in 1.6, replaced with something else) |
| 17:02.49 | raden_work | <PROTECTED> |
| 17:03.01 | raden_work | hold is the only time the status is changing in asterisk ???? |
| 17:03.05 | WIMPy | Jup. That's one for sip.conf. |
| 17:03.15 | manxpower | raden_work: YES! OBVIOUSLY YOU SCREWED UP YOUR CONFIG |
| 17:03.25 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 17:03.30 | manxpower | Now, if you pastebin your sip.conf maybe someone here can help you. |
| 17:03.37 | raden_work | manxpower, give me a moment |
| 17:03.50 | manxpower | raden_work: I didn't say ME, I said SOMEONE. |
| 17:03.51 | p3nguin | Make sure you have call-limit set on the peers. |
| 17:04.07 | raden_work | p3nguin, why would that make asterisk not change call status ???? |
| 17:04.07 | WIMPy | raden_work: Do you have call-limit? |
| 17:04.12 | raden_work | call-limit=5 |
| 17:04.14 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 17:04.29 | raden_work | do they need to be in call groups ???? |
| 17:04.41 | WIMPy | no |
| 17:04.41 | p3nguin | no |
| 17:05.12 | manxpower | raden_work: what version of Asterisk? |
| 17:05.20 | mutax | WIMPy: I still only get garbled audio -.- as if the samples are feed with 16 bit but the adapter only wants 8 or sth. :( |
| 17:05.27 | raden_work | 1.6 |
| 17:05.34 | manxpower | raden_work: FAIL. Try again. |
| 17:05.45 | WIMPy | mutax: Wrong speed? |
| 17:05.46 | manxpower | 1.6 is not an asterisk version. |
| 17:05.52 | raden_work | 1.6.0.10 |
| 17:06.13 | raden_work | that better ? |
| 17:06.34 | mutax | WIMPy: not sure. The adapter manual says they are using alaw an ptmp, misdn_info or sth. also says ptmp is in use. so I think the audio is a problem. |
| 17:06.39 | mutax | maybe in the kernel, dunno. |
| 17:06.47 | mutax | I'll look into mISDN code now |
| 17:07.09 | p3nguin | Is it possible that the phone (SIP/101) does not correctly inform Asterisk of its state? |
| 17:07.28 | hardwire | isn't the state defined from asterisk.. not from phone? |
| 17:07.34 | WIMPy | mutax: You usually don't have much choice the so no problem, either. Usually. |
| 17:07.40 | p3nguin | I don't know. |
| 17:07.54 | mutax | WIMPy: what? |
| 17:08.02 | WIMPy | thinks hardwire is right |
| 17:08.09 | manxpower | raden_work: why are you not using the latest 1.6.0.x? |
| 17:08.10 | raden_work | p3nguin, how does asterisk monitor it |
| 17:08.12 | hardwire | WIMPy: suck up |
| 17:08.15 | hardwire | :P |
| 17:08.17 | WIMPy | mutax: Audio format. |
| 17:08.24 | raden_work | manxpower, cause this Fing company never give me time todo anything like that |
| 17:08.35 | p3nguin | I don't use BLF (I have Cisco phones), and Asterisk thinks my phone is State:InUse even when I have it on hold. |
| 17:08.35 | hardwire | raden_work: try cloning. |
| 17:08.36 | mutax | WIMPy: yea, but hardware initialisation of the dsp seems to be broken or sth. |
| 17:08.53 | hardwire | p3nguin: it is in use |
| 17:09.02 | WIMPy | manxpower: What hardware? |
| 17:09.16 | p3nguin | His shows "Hold" as the state. |
| 17:09.26 | manxpower | raden_work: Do you have limitonpeer=yes? |
| 17:09.40 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 17:09.44 | manxpower | WIMPy: what hardware what? |
| 17:10.03 | raden_work | manxpower, yes |
| 17:10.17 | WIMPy | Err, sorry, meant mutax |
| 17:10.25 | manxpower | raden_work: then you need to go read UPGRADE-1.6.txt again and again until you stop using deprecated options. |
| 17:11.01 | kazaa_lite | ok... now i have voicemail configured:D I want to save my voicemail in slin16 audio..... what audio formet should i specify for this in voicemail.conf file? |
| 17:11.15 | p3nguin | hardwire: No explanation for that? |
| 17:11.28 | hardwire | p3nguin: actually I have to read back a bit |
| 17:11.40 | manxpower | Line 197 of UPGRADE-1.6.txt: The "limitonpeer" configuration option is now renamed to ....... |
| 17:12.54 | manxpower | p3nguin / hardware I suspect it depends on your version of Asterisk. 1.4.x didn't have many phone states supported, so most were just "in use" |
| 17:13.00 | *** join/#asterisk linuxcentos (~linuxcent@rhelbox.uio.no) |
| 17:13.28 | p3nguin | Ah, so his 1.6.x having Hold is new. |
| 17:14.51 | raden_work | omfg |
| 17:14.53 | raden_work | limitonpeer |
| 17:14.59 | raden_work | not limitonpeers FUCKING BOOK ! |
| 17:15.09 | p3nguin | does not exist in 1.6.x |
| 17:15.11 | hardwire | language.. you bastard.. language! |
| 17:15.14 | raden_work | it working ! |
| 17:15.21 | manxpower | raden_work: Perhaps you should read the docs that come with Asterisk |
| 17:15.31 | manxpower | or the sample config files |
| 17:15.47 | raden_work | trying to find them |
| 17:16.08 | *** join/#asterisk dan__t (vpn@vpn.withparity.net) |
| 17:16.09 | manxpower | raden_work: they are in the fricken doc/ directory of the tarball. Yes, I know nobody thinks to look there. |
| 17:16.11 | dan__t | Good morning. |
| 17:16.31 | p3nguin | I looked in the sample sip.conf for 1.6.something and "limiton" isn't found. |
| 17:16.35 | manxpower | Granted, for some reason the UPGRADE* files are in the root directory of the tarball. |
| 17:16.38 | hardwire | now I'm going to say "frick" all day... |
| 17:16.58 | seanbright | and i'll call you elliot |
| 17:17.01 | raden_work | manxpower, im trying to figure out where the morons moved everything I had neatly organized in the asterisk directory I have not touched this for 3 months they tried doing it themselves and i had to restart from backup files from over 6 months ago so please gimme a lil slack |
| 17:17.48 | p3nguin | That's a new Linux rapper's name: Li'l Slack |
| 17:18.34 | dan__t | Trying to settle a little bet here, and I figured this was the best place to ask. Is there a difference between a DID used for VoIP services in an originating manner, or a DID used on traditional POTS in an originating manner? Can the destination detect which is which? Is there anything to detect? Its the same type of number, same everything, it just happens to be originated differently somehow by the CLEC or ILEC? |
| 17:19.26 | p3nguin | I think it originates in the same way, since even VoIP carriers get the calls off the PSTN. |
| 17:19.36 | dan__t | Yeah, they'd have to |
| 17:20.07 | raden_work | there is nothing in changelog about limitonpeer |
| 17:21.29 | raden_work | manxpower, can u point me in the direction you were getting at cause in the docs im not finding anything with limit at all |
| 17:22.03 | p3nguin | As I mentioned, "limiton" wasn't found in the sample sip.conf, so they have obviously changed it to something else. |
| 17:22.43 | p3nguin | manxpower was able to give you the line number where the change is documented. |
| 17:22.48 | WIMPy | raden_work: You have been told which file. Even the line number. |
| 17:23.49 | bmoraca_work | dan__t: there's no difference whatsoever. there are certainly ways to tell, but they're not going to waste resources looking up every single number |
| 17:24.03 | dan__t | Right. |
| 17:24.41 | raden_work | <PROTECTED> |
| 17:24.55 | raden_work | ^^^ line 197 UPGRADE-1.6.txt |
| 17:25.29 | raden_work | why couldnt someone just tell me counteronpeer |
| 17:25.31 | raden_work | jesus |
| 17:26.15 | patrb | raden_workits not anyones job to tell you anything |
| 17:28.08 | raden_work | patrb, is that the most intelligent thing you could come up with ??? did I ever say it was ??? I think its childish that someone who knows the answer to something can't just be like they changed it to counteronpeer in 1.6 |
| 17:28.22 | raden_work | instead of wasting my freaking time .... |
| 17:28.44 | patrb | raden_work: I think its unprofessional that you work on asterisk servers but dont know how to read a damn doc...then come into #asterisk and demand answers |
| 17:29.36 | WIMPy | raden_work: If you had read the docs before, you wouldn't wast other people's time. And not yours, either. |
| 17:29.55 | bmoraca_work | wow, you guys need to chill out |
| 17:30.19 | raden_work | patrb, did not demand anything and i read plenty of documentation no where in any documentation did I see anything saying this was changed in 1.6 . and for the record both of the books I have cover 1.6 so there is a error in documentation but of course everyone feels a need to be hormonal |
| 17:30.23 | bmoraca_work | people help out as they can. it's entirely possible that someone didn't know that answer off the top of their head. i know that I didn't. |
| 17:30.40 | p3nguin | Maybe the people who know think that you are wasting their time by not reading the doc yourself. |
| 17:30.53 | Katty | hi |
| 17:30.54 | raden_work | Its freaking ridiculous the amount of unprofessional conduct in here its basically your stupid RTFM ! |
| 17:31.15 | dan__t | Maybe you should RTFM. |
| 17:31.15 | patrb | raden_work: lol, you're an idiot |
| 17:31.19 | p3nguin | That's how most of us got where we are. |
| 17:31.29 | raden_work | p3nguin, I have read it about 6 months ago, don't rember lil things like that |
| 17:31.38 | dan__t | No one held my hand and wiped my ass along the way. SUre, it was a rough start, but it taught me to think. |
| 17:31.52 | manxpower | raden_work: I DOWNLOADED 1.6.0.10 and looked. Either you don't have 1.6.0.10 or something else is wrong |
| 17:31.57 | chuckf | hi Katty |
| 17:32.00 | raden_work | If I had 1.4 it would not be a problem ! |
| 17:32.03 | dan__t | p3nguin, remember that dialer thing I was working on some months ago. |
| 17:32.10 | raden_work | manxpower, its working now |
| 17:32.13 | dan__t | I ended up getting it to work, I wanted to thank you again for your help. |
| 17:32.13 | raden_work | everything is working |
| 17:32.32 | p3nguin | dan__t: Honestly, no. I've seen a lot of people working on some unusual things, so I don't remember all of them. |
| 17:32.33 | raden_work | 96% of what I do i never get help for |
| 17:32.33 | manxpower | raden_work: until the next problem and then we will go thru this whole thing again. |
| 17:32.37 | dan__t | I'm sure you do. |
| 17:32.40 | dan__t | Regardless; thanks. |
| 17:33.11 | raden_work | am i suppose to reread the book and all the upgrade docs everytime I need todo something |
| 17:33.12 | raden_work | ? |
| 17:33.23 | patrb | raden_work: in your case: yes |
| 17:33.29 | raden_work | I followed tutorials read book etc.... |
| 17:33.47 | dan__t | Yep. |
| 17:33.58 | patrb | raden_work: jokes aside...yes, everyone should always read the docs |
| 17:34.07 | raden_work | patrb, your one of them whimpy pencil pushers that really doesn't have to work much are you ? |
| 17:34.29 | patrb | raden_work: im an asterisk admin...so yeah I dont usually have a lot to do |
| 17:34.52 | raden_work | patrb, yea so sit and read all your docs |
| 17:35.12 | patrb | raden_work: I do...and thats why I dont have problems on my machines |
| 17:35.41 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:35.46 | patrb | raden_work: i also lurk the hell out of this channel to learn more |
| 17:35.53 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 17:36.32 | raden_work | I do accounting , electronics , purchasing , im on the road as a service tech 75% of the time amongst a mist of other things they alot no time for these things thats why i dont have time to read docs everytime. |
| 17:36.58 | patrb | raden_work: i dont care. just dont come in here and get pissed that people dont answer your questions |
| 17:37.12 | patrb | raden_work: no one is getting paid to be in here |
| 17:37.16 | bmoraca_work | figured out why one of my servers keeps rebooting...BAD POWER! |
| 17:37.18 | bmoraca_work | uhg |
| 17:37.28 | bmoraca_work | well, i'm off to fix it |
| 17:37.38 | WIMPy | If you don't have time to do the job yoursef, pay someone else to do it. |
| 17:37.38 | raden_work | well they have time to put me down... tell me to read ... they know the answer why everyone have to be a dick about it!@ |
| 17:37.55 | raden_work | WIMPy, im the guy that gets paid to do it |
| 17:38.11 | dan__t | Sounds like a shitty deal to me. |
| 17:38.14 | raden_work | this company is the absolute worst for organization |
| 17:38.18 | raden_work | dan__t, very |
| 17:39.05 | WIMPy | raden_work: Yes, but you want others to do it. That obviousely doesn't work out. |
| 17:39.06 | dan__t | Know what I did at work yesterday? Played baseball in the office with cardboard and bouncy balls :/ |
| 17:39.14 | Naikrovek | nerf guns are funner |
| 17:39.16 | raden_work | WIMPy, show me where i asked anyone todo anything |
| 17:39.17 | Naikrovek | more fun* |
| 17:39.34 | patrb | <raden_work> why couldnt someone just tell me counteronpeer |
| 17:39.34 | patrb | <raden_work> jesus |
| 17:39.44 | raden_work | the whole issue here is alls manxpower had to say is its counteronpeer not limitonpeer in 1.6 jesus |
| 17:39.50 | WIMPy | thanks patrb |
| 17:40.00 | raden_work | thats a simple question |
| 17:40.07 | raden_work | if someone knows why cant they say he its this |
| 17:40.24 | seanbright | because it promotes support vampirism |
| 17:40.30 | Naikrovek | eh |
| 17:40.34 | Naikrovek | we're in here to answer questions |
| 17:40.36 | Naikrovek | i am anyway |
| 17:40.39 | raden_work | take the time to lookup the doc look for what i need in the doc and give me a line number etc... but cant just say hey its this |
| 17:40.39 | seanbright | not me |
| 17:40.43 | raden_work | makes no sense !!! |
| 17:40.43 | seanbright | i'm here for the ladies |
| 17:40.45 | p3nguin | enabler! |
| 17:41.00 | raden_work | i help people and answer things directly when I can |
| 17:41.30 | leifmadsen | kazaa_lite: have you just tried adding "slin16" to the list of formats in voicemail.conf? |
| 17:41.36 | patrb | raden_work: I dont think anyone knew that answer, and then your attitude turned them off from helping |
| 17:41.39 | p3nguin | Hmm, what's going to be for lunch today? |
| 17:41.48 | raden_work | patrb, yup all me again :) |
| 17:41.55 | p3nguin | Maybe chicken fries. |
| 17:42.00 | leifmadsen | kazaa_lite: or rather sln16 |
| 17:42.04 | Naikrovek | ooh chicken fries |
| 17:42.13 | seanbright | kfc double down |
| 17:42.15 | kazaa_lite | leifmadsen: yes.... and i got this message: no such file format slin16 |
| 17:42.17 | hardwire | eww |
| 17:42.20 | leifmadsen | format=wav49|gsm|wav |
| 17:42.28 | leifmadsen | kazaa_lite: you tried sln16 ? |
| 17:42.35 | kazaa_lite | leifmadsen: lemme try sln16 |
| 17:42.54 | hardwire | leifmadsen: is 1.6.2 meetme 16khz capable? |
| 17:42.58 | hardwire | ponders |
| 17:42.59 | chuckf | raden_work: someone answered your question and you didn't like the way it was answered, I say demand a refund on the consulting costs! |
| 17:43.00 | leifmadsen | hardwire: good question :) |
| 17:43.01 | [TK]D-Fender | hardwire: no |
| 17:43.14 | hardwire | [TK]D-Fender: :| |
| 17:43.21 | l2trace99 | anyone know how I can change the leg chanspy whispers to ? |
| 17:43.23 | [TK]D-Fender | hardwire: *'s mixing core is 8khz. |
| 17:43.24 | hardwire | leifmadsen: :| |
| 17:43.44 | hardwire | [TK]D-Fender: hmm.. I thought the bridge stuff supported arbitrary stuffins |
| 17:43.46 | leifmadsen | hardwire: sounds like something the new bridging API may or may not resolve? Which means you'd have to use app_confbridge potentially |
| 17:43.55 | hardwire | yeh |
| 17:43.59 | hardwire | I'd be down with that. |
| 17:44.03 | leifmadsen | (since app_confbridge is the only application that uses the new bridging API) |
| 17:44.05 | hardwire | wonders how he can help. |
| 17:44.11 | leifmadsen | (as I learnt only this morning from file! :)) |
| 17:44.20 | hardwire | sneaky little file |
| 17:44.35 | kazaa_lite | leifmadsen: it works:D |
| 17:45.00 | seanbright | l2trace99: i guess you'd have to call ChanSpy on the other channel |
| 17:45.05 | leifmadsen | kazaa_lite: I just looked in the formats/ directory and saw the file name was format_sln16 |
| 17:45.54 | l2trace99 | any idea of how I can get the channel id of the other channel ? |
| 17:46.03 | seanbright | l2trace99: ChanSpy(${BRIDGEPEER}...) |
| 17:46.05 | mutax | WIMPy: great. software suspend produces a kernel oops with htf_usb :) this really is a nice peace of software ;P |
| 17:46.27 | l2trace99 | damn faster than I could type the question |
| 17:46.27 | p3nguin | l2trace99: Spy the other channel. |
| 17:46.38 | seanbright | l2trace99: i knew it was coming :-) |
| 17:46.53 | kazaa_lite | leifmadsen: ahhh... never noticed there is formats directory as well |
| 17:46.54 | mutax | WIMPy: what kernel version do you use? |
| 17:47.06 | leifmadsen | kazaa_lite: all modules come from somewhere :) |
| 17:47.06 | seanbright | l2trace99: actually, that won't work. |
| 17:47.19 | WIMPy | mutax: I don't think I had one, but it definitely didn't work any more after a suspend. |
| 17:47.22 | seanbright | l2trace99: since you aren't part of the bridge |
| 17:47.52 | l2trace99 | but if I answer before I outward dial |
| 17:47.59 | l2trace99 | could I grab it ? |
| 17:48.16 | l2trace99 | would the channel id of that leg follow ? |
| 17:48.18 | seanbright | how are you calling ChanSpy now? |
| 17:48.38 | l2trace99 | setting a SPYGROUP |
| 17:48.45 | WIMPy | mutax: 2.6.33-rc5 on the big one, 2.6.31 on the netbook |
| 17:48.46 | l2trace99 | var and keying on that |
| 17:49.12 | l2trace99 | this box sits inbetween |
| 17:49.33 | l2trace99 | none of the calls are registered to it |
| 17:50.02 | l2trace99 | s/calls/phones/ |
| 17:50.06 | seanbright | well, long story short: if you want to whisper to the other side you need to be ChanSpy'ing on the other side |
| 17:50.16 | seanbright | now you do that... i'm not 100% |
| 17:50.21 | seanbright | s/now/how/ |
| 17:52.23 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 17:52.33 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 17:55.35 | kazaa_lite | leifmadsen: true:) |
| 17:56.24 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 17:56.29 | mutax | WIMPy: hmm. strange. this audio problem suxxs |
| 17:57.43 | WIMPy | mutax: Is only the audio problem left? |
| 17:58.03 | hardwire | problems hav a sux |
| 17:58.36 | mutax | WIMPy: well, sort of. I am testing with lcr all the time now, because I just make no sense hooking this up to asterisk |
| 17:58.48 | *** join/#asterisk JKac3BEq (~JKac3BEq@pl2.haspere.com) |
| 17:59.04 | mutax | WIMPy: it might be a probmel with the dsp part |
| 17:59.09 | WIMPy | mutax: Do inbound calls work then? |
| 17:59.57 | mutax | WIMPy: I dont think so but I have no way to test it right now |
| 18:00.17 | mutax | WIMPy: I dont wanna set up a voice boy in lcr ;) |
| 18:00.19 | WIMPy | mutax: Have you tried another cable? |
| 18:00.29 | mutax | for isdn? sure |
| 18:00.37 | mutax | thats not the problem. |
| 18:00.55 | mutax | and the audio sounds digitally distorted, not analog |
| 18:01.10 | mutax | its definitily a software problem on my notebook |
| 18:01.29 | WIMPy | Well, how could there possibly be an analogue distortion? |
| 18:02.24 | WIMPy | I think it might be worth while to try another machine then. |
| 18:02.46 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 18:04.32 | mutax | WIMPy: the analogue distortion? between the cable modem and the isdn converter :o) |
| 18:05.14 | *** part/#asterisk TommyBotten (tommy@broken.pipe.no) |
| 18:05.19 | WIMPy | mutax: Or use something else for testing than that adaptor. |
| 18:05.57 | mutax | yes - as soon as I find a person with ISDN -.- they are rare |
| 18:05.58 | *** part/#asterisk manxpower (~ewieling@216.186.151.147) |
| 18:06.36 | WIMPy | mutax: But you have ISDN terminal(s). You could try it the other way round. |
| 18:08.02 | mutax | humm? no, I dont have an isdn phone |
| 18:08.14 | mutax | I could in fact setup another machine with asterisk and isdn card |
| 18:08.24 | WIMPy | What do you have connected to that adaptor then? |
| 18:08.48 | mutax | setup is cable modem -> analogue lines -> isdn adaptor -> usb isdn ta |
| 18:09.09 | WIMPy | Oh, you only have it for the TA? |
| 18:09.15 | mutax | jupp |
| 18:09.21 | WIMPy | Bad luck. |
| 18:09.43 | WIMPy | I think you should find (or make) a third device. |
| 18:09.52 | mutax | interesting: when I switch audio to ulaw nothing changes! |
| 18:10.08 | mutax | maybe the kernel driver sends ulaw all the time |
| 18:11.21 | *** join/#asterisk cnu (~cnu@the.ultimate.lamer.la) |
| 18:11.51 | redax | WIMPy: reverted the driver to wcb4xxp as not the driver was the problem. btw, I'll play with LCR, seems like promising. |
| 18:12.05 | redax | WIMPy: cabling guy mixed the S0 wires :-( |
| 18:13.02 | *** join/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 18:13.05 | WIMPy | redax: Electricians? Well, the nice thing is that both parts can work independantly. |
| 18:13.51 | redax | WIMPy: the bad thing is if you mix 1wire from pair-A to Pair-B the layer1 goes up for 5-6 seconds |
| 18:14.05 | redax | after that it goes to YELow alarm, and again goes OK |
| 18:14.27 | redax | so hell's though it is because of the mixed wires... |
| 18:14.53 | *** join/#asterisk Tim_Toady (~moi@188.4.0.11.dsl.dyn.forthnet.gr) |
| 18:15.02 | WIMPy | Just four duo-LEDs can be pretty helpfull |
| 18:15.04 | redax | as we're interconnecting a Siemens HiCom 330 and asterisk via 2x WC B410P |
| 18:17.26 | redax | so we can say if anyone needs mISDN, then chan_lcr should work rock stable, right? |
| 18:17.50 | redax | not as stable as mISDN worked earlier with chan_misdn + asterisk 1.4 |
| 18:18.19 | *** join/#asterisk JKac3BEq (~JKac3BEq@pl2.haspere.com) |
| 18:18.20 | coppice | stable and misdn in the same sentence. weird |
| 18:18.24 | WIMPy | It has been working very well for several month now. Except for a little bug in SendText, that caused it to lock up. |
| 18:18.50 | WIMPy | It has been mor stable than chan_misdn. |
| 18:19.09 | *** join/#asterisk mlr78 (~mratliff@74.5.197.138) |
| 18:19.31 | *** join/#asterisk xpot-mobile (~james@66.60.101.91) |
| 18:20.50 | redax | WIMPy: I think I'm gonna be happy without sending pervert messages to the user's phone display :D |
| 18:21.29 | WIMPy | Well I used it in responses to forwarding inquiries. Took me some time to find that bug. |
| 18:21.53 | WIMPy | However even after fixing it, there's still no text appearing on the phones. |
| 18:22.25 | redax | anyway does SendText() working for other phones than isdn ? |
| 18:22.53 | WIMPy | It works with SIP, but NOT chan_lcr. |
| 18:23.50 | thehar | leifmadsen: i love when you touch my ticket |
| 18:24.23 | leifmadsen | thehar: o.O eeep! |
| 18:24.37 | thehar | leifmadsen: lol #0017171 |
| 18:24.43 | hardwire | fancy |
| 18:24.48 | thehar | ikr |
| 18:24.58 | thehar | giggles |
| 18:26.35 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 18:26.48 | thehar | leifmadsen: i was joking.. long time :) how's it goin |
| 18:27.46 | leifmadsen | haha not bad :) |
| 18:28.04 | thehar | sweet. |
| 18:28.09 | thehar | Astricon DC? |
| 18:28.15 | thehar | can't decide if i should attend or not this year |
| 18:28.22 | leifmadsen | YOU SHOULD! |
| 18:28.24 | leifmadsen | I'll be there! |
| 18:28.27 | *** join/#asterisk blaines (~blaines@ip70-190-67-139.ph.ph.cox.net) |
| 18:28.31 | leifmadsen | there is no other reason that you should need |
| 18:28.35 | thehar | lol |
| 18:28.43 | WIMPy | leifmadsen: I'm not so familiar with the issue tracker. Is there anything I have to do to get issue 16157 further on? |
| 18:28.51 | thehar | but you're always so busy |
| 18:29.04 | leifmadsen | M16157 |
| 18:29.06 | MuffinMan | [ready for testing] [Asterisk] Applications/app_dial 0016157: [patch] Wrong cause value for 'answered elsewhere' reported by wimpy https://issues.asterisk.org/view.php?id=16157 |
| 18:29.53 | leifmadsen | WIMPy: eep! That is old! I will get it put into the internal tracker so it can get prioritized and some resources assigned to it |
| 18:29.57 | leifmadsen | thanks for the pointing it out to me! |
| 18:30.01 | *** join/#asterisk kartikm (~koolkarti@117.199.118.28) |
| 18:30.30 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 18:30.53 | WIMPy | leifmadsen: Ok. Just wanted to make sure it isn't lost. |
| 18:31.10 | leifmadsen | WIMPy: np... sometimes things do get lost and we just need a little poke |
| 18:33.00 | WIMPy | So much hassle for just a little 7-Bit value... |
| 18:35.45 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:37.29 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 18:38.26 | *** join/#asterisk friartuck (~pmccary@66.162.90.56) |
| 18:40.13 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 18:40.15 | Assid | heya |
| 18:40.47 | leifmadsen | WIMPy: I'll see if I can get someone to look at it |
| 18:40.52 | Assid | anyone know whats up with ipkall.. i want to register.. cause i misspelled my address and now it keeps saying ive registere |
| 18:41.21 | l2trace99 | seanbright: talking about making harder than what it it |
| 18:41.25 | l2trace99 | is |
| 18:41.58 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 18:42.24 | WIMPy | leifmadsen: What would the official way have been to get things back out of the attic, if not here? |
| 18:42.59 | leifmadsen | WIMPy: probably just pinging the issue -- unfortunately with limited resources and over 600 open issues, there is only so much attention to go around |
| 18:43.09 | leifmadsen | WIMPy: I'm trying to see if I can somehow fast track it though |
| 18:43.14 | l2trace99 | seanbright: there was an issue with chanspy and wisper for 1.6.1.6 |
| 18:43.51 | l2trace99 | seanbright: I upgraded to 1.6.2.6 all works |
| 18:44.12 | WIMPy | leifmadsen: Yes, sure. But at least that one is trivial. |
| 18:44.14 | leifmadsen | that's a major version chage :) |
| 18:44.26 | leifmadsen | WIMPy: yep for sure, but still requires resources to review and commit |
| 18:44.50 | l2trace99 | so I figured it had to be a bug because I never do anything wrong |
| 18:44.51 | l2trace99 | ;) |
| 18:45.03 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 18:45.21 | l2trace99 | the problem is never on this side of keyboard |
| 18:45.51 | seanbright | l2trace99: glad you got it working |
| 18:46.06 | l2trace99 | thanks I thought I was going crazy |
| 18:46.21 | WIMPy | leifmadsen: Yes, I do understand. As I said I just wanted to make sure it's not lost and hopefully makes it upstream some time. |
| 18:48.14 | leifmadsen | WIMPy: no worries, thanks for pinging me about it! |
| 18:48.22 | *** join/#asterisk Pazzo (~ugelt@sadsl-246059.rol.raiffeisen.net) |
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| 19:17.40 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 19:17.49 | cusco | ih |
| 19:18.10 | cusco | whenever asterisk dials a peer, goes trough Set(SPYGROUP=${EXTEN}); |
| 19:18.41 | cusco | now for some reason Chanspy(,bqg(${EXTEN:3:3})) ain't working |
| 19:19.09 | cusco | Executing [007617@agents:5] ChanSpy("SIP/280-00000404", ",bqg(617)") in new stack |
| 19:19.32 | cusco | [Apr 13 19:31:02] VERBOSE[7177] pbx.c: -- Executing [617@agents:1] Set("Local/617@agents-4e7c;2", "SPYGROUP=617") in new stack |
| 19:20.04 | cusco | now the peer 617 is remote on a iax trunk... |
| 19:23.20 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 19:25.15 | *** part/#asterisk kazaa_lite (~eddie@78-86-126-14.zone2.bethere.co.uk) |
| 19:26.28 | *** join/#asterisk pawz (~pawz@ppp118-208-100-151.lns20.bne4.internode.on.net) |
| 19:27.04 | l2trace99 | cusco: any chance you seting spygroup 2 ? |
| 19:28.29 | cusco | spygroup 2? |
| 19:28.39 | ba1dy | any german asterisk profi here which i am allowed to talk in a query? ;) |
| 19:28.55 | l2trace99 | s/2/twice |
| 19:29.20 | cusco | where should I set it next? |
| 19:29.32 | *** join/#asterisk awkfu (~awkfu@pixpat0.coremetrics.com) |
| 19:30.14 | WIMPy | ba1dy: You could try to open a german channel. |
| 19:30.15 | Naikrovek | ah i love irc |
| 19:30.21 | Naikrovek | wish i could just sit here all day and learn |
| 19:30.22 | l2trace99 | it looks like when it calls the SPYGROUP var it is looking for a SIP channel |
| 19:30.50 | l2trace99 | but it is dialing out through the local |
| 19:31.59 | ba1dy | WIMPy: asterisk.de already exist but there is nobody ;) |
| 19:32.05 | l2trace99 | so I suspect that SPYGROUP is not being set on the correct channel |
| 19:32.15 | raden_work | is threre a way if phone status DND it will send to another extension |
| 19:33.02 | cusco | l2trace99: I don't think ir is looking for a sip channel |
| 19:33.11 | cusco | l2trace99: sip/280 its me making the call |
| 19:33.36 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 19:34.22 | cusco | so you're saying I can just Chanspy(Local/${EXTEN}) ? |
| 19:34.30 | [TK]D-Fender | raden_work: Depends if the phone responds with a code that can be identified as DND. There is no standard |
| 19:35.40 | cusco | that would not work |
| 19:36.25 | l2trace99 | cusco: no I am saying that it sounds like you calling SET(SPYGROUP=${EXTEN3:-3}) on 2 separate channels |
| 19:36.43 | l2trace99 | and you are getting the one you want |
| 19:36.50 | l2trace99 | or expect |
| 19:38.04 | cusco | on _XXX => { I have: Set(SPYGROUP=${EXTEN}); |
| 19:38.14 | *** part/#asterisk mmoreno80 (~matt@190.190.47.146) |
| 19:38.26 | cusco | and on _007XXX => { there is: Chanspy(,bqg(${EXTEN:3:3})); |
| 19:38.34 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
| 19:38.48 | cusco | so I would get to spy on bridged channels.. |
| 19:39.09 | cusco | the dial is: Dial(IAX2/primario-iax-cov@covilha-iax/${EXTEN}@agents,7,rktgA(${NewCallMsg})); |
| 19:39.47 | raden_work | <PROTECTED> |
| 19:39.47 | raden_work | <PROTECTED> |
| 19:39.47 | raden_work | <PROTECTED> |
| 19:39.59 | raden_work | [TK]D-Fender, thats what happens when DND SET |
| 19:40.23 | [TK]D-Fender | raden_work: And what do *'s return values say? |
| 19:40.38 | [TK]D-Fender | raden_work: do you get that if all call appearances are taken up as well? |
| 19:40.54 | *** join/#asterisk Brixius (~Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 19:41.52 | raden_work | [TK]D-Fender, im not following you ... |
| 19:43.47 | [TK]D-Fender | raden_work: What's hard to follow? look at ${DIALSTATUS} etc for these calls and see if the answer is UNIQUE. If the phone is alerady busy with REAL CALLS (to the max) will it return the same? |
| 19:44.01 | *** join/#asterisk frek818 (~herman@rrcs-76-79-184-51.west.biz.rr.com) |
| 19:44.38 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:44.38 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 19:47.06 | *** join/#asterisk pawz (~pawz@ppp118-208-100-151.lns20.bne4.internode.on.net) |
| 19:47.12 | raden_work | gotcha |
| 19:47.18 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:48.08 | Skeeter- | whos got custom ringtone working with polycoms? |
| 19:49.17 | *** join/#asterisk fuxu2 (iconicflux@www.kevinlynn.com) |
| 19:51.23 | bmoraca_work | i do |
| 19:52.21 | Naikrovek | i do too |
| 19:52.38 | Skeeter- | how |
| 19:52.53 | Skeeter- | i tried many sox commands and so far withouth success |
| 19:53.13 | Skeeter- | 3-4 months ago, i had the perfect working command, i lost it.. |
| 19:53.41 | bmoraca_work | i don't recall needing anything special |
| 19:54.03 | bmoraca_work | however, i would check the polycom administrator guide to get the exact specs of the file. |
| 19:54.14 | WIMPy | ba1dy: There is an example including dtmf response in that document. What elese do you need? |
| 19:54.42 | bmoraca_work | aside from that, are you sure that the phone is downloading the file? is your sip.conf file properly set up? how do you know it's a format issue? |
| 19:54.49 | Naikrovek | Skeeter-: there's no special command you just create a file that has that particular section of the config in it, and tell your [mac].cfg to load that file |
| 19:55.11 | ba1dy | WIMPy: in which? from netways? |
| 19:55.15 | Skeeter- | i did |
| 19:55.36 | WIMPy | ba1dy: You only showed me one. |
| 19:56.08 | bmoraca_work | Skeeter-: show us that the phone is downloading the file and simply refusing to play it |
| 19:56.12 | ba1dy | mhh tell we where.. cant find an example |
| 19:57.02 | WIMPy | ba1dy: Page 21 |
| 19:57.03 | Skeeter- | well, within the file i set the blabla.wav but when i wanna configure it within the phone menu, it doesnt sees it |
| 19:58.03 | bmoraca_work | well then your phone must be broken because you obviously have everything configured properly |
| 20:02.18 | Skeeter- | trying sth, if it doesnt work, ill pastebin the files |
| 20:03.34 | raden_work | why when I do exten => 0,1,dial(SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/120,20) |
| 20:03.35 | raden_work | <PROTECTED> |
| 20:04.01 | raden_work | 103 is directed to my cell in astDB if i dial 103 goes to cell |
| 20:04.09 | raden_work | but when its grouped like that wont work |
| 20:04.49 | Naikrovek | Skeeter-: what does the 'file' command return when you run it on that .wav |
| 20:04.59 | Naikrovek | Skeeter-: and what model phone and how big is the .wav |
| 20:04.59 | Skeeter- | http://pastebin.com/LtcJUVkc |
| 20:05.05 | Skeeter- | 550 |
| 20:05.17 | bmoraca_work | raden_work: because you're not telling it to. |
| 20:05.37 | raden_work | ? |
| 20:05.42 | Skeeter- | dist.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz |
| 20:05.42 | bmoraca_work | raden_work: astdb isn't some magical land where the dialplan knows to look for everything. you have to TELL it to look there. |
| 20:05.59 | raden_work | interesting ... |
| 20:06.01 | Skeeter- | 84k big |
| 20:06.17 | raden_work | sip/103 looks there but not (SIP/101&SIP/102&SIP/103&SIP/104&SIP/105&SIP/120,20) |
| 20:06.19 | Naikrovek | Skeeter-: the custom ringer goes in the sip.cfg file, not phone1.cfg. see http://pastebin.com/1qpijUSM |
| 20:06.25 | raden_work | bmoraca_work, how would i go about doing that ? |
| 20:06.38 | Skeeter- | i tried in sip.cfg, that was my last attempt |
| 20:06.44 | Skeeter- | will give another try |
| 20:07.12 | bmoraca_work | raden_work: some logic that looks in astdb and determines whether or not to forward it and where to forward it to |
| 20:07.16 | Naikrovek | Skeeter-: create a NEW file, put http://pastebin.com/1qpijUSM in it |
| 20:07.39 | Naikrovek | Skeeter-: then reference that file from 0004f2123456.cfg or whatever your MAC address is |
| 20:07.47 | Naikrovek | Skeeter-: put it BEFORE sip.cfg in the list of config files |
| 20:08.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:09.30 | Naikrovek | Skeeter-: the [mac].cfg should look something like http://pastebin.com/KCzdGMBt |
| 20:09.43 | Naikrovek | note the ringer_24.cfg in the list of config files |
| 20:09.52 | Naikrovek | then refer to my previous pastebin for those file contents |
| 20:09.55 | mutax | WIMPy: you know what? I wanted to test with another PC. the first one need hours to compile the kernel for the usb patch (old notebook, so no pci) the other one with pci bus: I pulled it out of the rack, installed the isdn card, put it back in, reconnected all wires, bootet it up: cmos battery bad. after setting the time: kernel oopses during boot -.- . I hereby declare this day to be one of the worst days in my life for weeks! *gnargh* [and my table |
| 20:09.57 | Skeeter- | wow, u put a lot of configs file |
| 20:10.04 | Naikrovek | Skeeter-: MUCH easier that way |
| 20:10.13 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:10.14 | Skeeter- | i perfectly understand it |
| 20:10.19 | Naikrovek | Skeeter-: and when i upgrade firmware i let it overwrite sip.cfg and phone1.cfg because i have these overrides |
| 20:10.31 | Skeeter- | nice |
| 20:10.37 | Skeeter- | very smart |
| 20:10.39 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 20:10.50 | Skeeter- | rebooting with your configs |
| 20:11.08 | fuxu2 | anyone here going to the atlanta asterisk user conference on Saturday? |
| 20:11.57 | Naikrovek | Skeeter-: watch the (t)ftp server logs to see if the .wav gets pulled down by the phone |
| 20:12.09 | Skeeter- | OH |
| 20:12.09 | Naikrovek | and i've noticed that if the .wav is more than about 300kb it won't work no matter what |
| 20:12.12 | Skeeter- | nice |
| 20:12.18 | Skeeter- | my polycom is frozen |
| 20:12.24 | Naikrovek | no it's not |
| 20:12.30 | Naikrovek | it's pulling down the .wav probably |
| 20:12.31 | Skeeter- | screen is |
| 20:12.47 | Naikrovek | wait a bit |
| 20:12.50 | Naikrovek | go take a leak or something |
| 20:12.55 | bmoraca_work | or failing to pull it down |
| 20:12.58 | Naikrovek | read newsweek from the lobby |
| 20:12.59 | Naikrovek | it'll come back |
| 20:13.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:13.07 | Skeeter- | lol |
| 20:13.07 | Naikrovek | bmoraca_work: that'll give another error quite promptly in my experience |
| 20:13.11 | bmoraca_work | check TFTP logs to see if you had a typo or if there's a privs issue |
| 20:13.12 | Naikrovek | 0x2 |
| 20:13.23 | Naikrovek | yeah typos are easy to introduce in there and will give odd errors |
| 20:13.28 | Skeeter- | previously i had like 5-6 custom ringtone, and it didnt that long, |
| 20:13.34 | bmoraca_work | Naikrovek: in the logs, yes, but at the screen itself, it'll look like it hangs |
| 20:13.52 | Naikrovek | really? on mine it gives error 0x2 and reboots in a loop |
| 20:13.57 | Naikrovek | i have dift. model phone though |
| 20:14.02 | bmoraca_work | Skeeter-: check your TFTP logs then. or the phone logs. wondering why it's hung doesn't do any good |
| 20:14.42 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 20:16.03 | Skeeter- | seems to get every files... |
| 20:17.55 | Skeeter- | log doesnt say it loads the .wav tho |
| 20:18.00 | Skeeter- | but it loads all files |
| 20:19.00 | Skeeter- | i guess ill have to get another polycom firmware to flash it |
| 20:19.17 | bmoraca_work | you've got a config issue then |
| 20:19.52 | Skeeter- | adding the sip2.cfg containing ONLY sampled blablabla makes the polycom hang on boot |
| 20:20.30 | *** join/#asterisk rare1980_ (~rare1980@115.186.26.210) |
| 20:20.33 | Skeeter- | i removed sip2.cfg from the loadup, and now it boots perfectly, SEEMS that when the ringtone works, and the configs work, it aactually makes the phone freeze |
| 20:20.40 | cusco | with realtime queues, in one of the queues periodic_announce doens't seem to be working.. hwo can I debug why? |
| 20:20.45 | Skeeter- | could be sth else |
| 20:20.48 | cusco | on a testing queue it works! |
| 20:20.56 | cusco | same ael, same mysql queue configuration |
| 20:21.01 | cusco | only context varies |
| 20:21.34 | Skeeter- | -rw-r--r-- 1 root root 84230 2010-04-13 16:00 Dist.wav |
| 20:22.08 | rare1980_ | ppl i need to know if i have paid someone in advance for some work and he took my money and ran away |
| 20:22.13 | rare1980_ | wht should i do then |
| 20:22.14 | rare1980_ | ? |
| 20:22.20 | Skeeter- | Naikrovek, whats the whole proccess you do in order to add a new custon ringtone |
| 20:22.30 | *** join/#asterisk jtodd (z5m5cgwtku@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 20:22.30 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 20:22.55 | raden_work | we have 5 phones ringing in the building sip101-105 the owner wants the call symbol on the caller id to show picked up if anyone picks up is that even possible ? / |
| 20:22.58 | rare1980_ | i paid some expert from this room |
| 20:23.13 | rare1980_ | his name is voipmonk.. |
| 20:23.23 | bmoraca_work | raden_work: "call symbol"? |
| 20:23.25 | rare1980_ | he took my money and he didn't finish my work |
| 20:23.29 | Skeeter- | rare1980_, he did that to me too |
| 20:23.38 | rare1980_ | wht |
| 20:23.40 | rare1980_ | really |
| 20:23.50 | Skeeter- | rare1980_, i gave him 200$, and nvr figured anything out |
| 20:23.50 | rare1980_ | did he pay you back |
| 20:23.56 | Skeeter- | nope |
| 20:24.01 | Skeeter- | work not ganratee |
| 20:24.01 | raden_work | bmoraca_work, there is a symbol on the phones caller id to show if the call was picked up or not per phone he wants me to sync them so if one phone picks up the call all the caller ids will show picked up |
| 20:24.09 | raden_work | I told him i doubt it was possible |
| 20:24.22 | raden_work | then i got our old panasonic phone system ( that never worked ) did that :( |
| 20:25.44 | bmoraca_work | raden_work: your terminology and phrasing don't make sense. i'm not understanding what you want. do you want busy-lamp field? |
| 20:27.08 | *** part/#asterisk l2trace99 (~jr@74.118.40.1) |
| 20:28.35 | raden_work | bmoraca_work, on the caller ID on the AASTRA 9133i phones when a call comes in it shows on the caller ID whether or not the call was answered well we have 5 phones he wants that stupid icon to be synced between the phones so he knows if calls are being missed |
| 20:28.58 | rare1980_ | skeeter- now what should we do about him |
| 20:29.28 | bmoraca_work | raden_work: not familiar with Aastra phones, but it seems to me that would be very phone-specific. it's certainly not a default feature of SIP. |
| 20:29.43 | raden_work | did not think sooo |
| 20:31.17 | Naikrovek | Skeeter-: i crop the audio file to a reasonable length (10-30 seconds) with Audacity, convert it to the proper format with sox, put it in the ftp directory, create a config file just for that ringtone, add that .cfg file entry to necessary config files, reboot necessary phones |
| 20:32.39 | Naikrovek | raden_work: get him a 550 or 650 with a sidecar so he can see not only if the call was picked up, but who picked it up |
| 20:32.46 | Naikrovek | polycom 550 or 650 |
| 20:32.51 | Naikrovek | maybe 450 if they have sidecars |
| 20:33.22 | raden_work | they record all that stuff ? |
| 20:33.49 | Skeeter- | i do the same here |
| 20:33.56 | Naikrovek | you know when a call rings everyone's phone because your phone is ringing. he can see who picks it up with the BLF stuff you got working on the Aastra today |
| 20:33.58 | Skeeter- | i save from mp3 to wav with audacity |
| 20:34.11 | bmoraca_work | raden_work: they don't "record" it, it's what's known as "Busy Lamp Field" |
| 20:34.39 | bmoraca_work | raden_work: it's set up with hints and speed dials on polycom phones. other phones set it up differently |
| 20:34.47 | Skeeter- | Anyone knows Voipmonk Real name? |
| 20:35.17 | raden_work | bmoraca_work, yea i have BLF working so he can see who on the phone and who picks up but he wants a log on the caller id like his panasonics had |
| 20:35.29 | raden_work | ready to make a CDR viewable via web browser |
| 20:35.37 | Naikrovek | Skeeter-: greg something |
| 20:35.39 | Qwell | Skeeter-: Greg, IIRC |
| 20:35.40 | bmoraca_work | raden_work: i don't believe that's possible. |
| 20:35.56 | Skeeter- | greg shido??? |
| 20:36.03 | Qwell | no |
| 20:36.06 | Naikrovek | raden_work: your boss is a micromanaging asshole and needs to be shunned and/or punished |
| 20:36.07 | raden_work | bmoraca_work, caller id thing ???? I know its not |
| 20:36.16 | raden_work | Naikrovek, i vote both |
| 20:36.20 | Naikrovek | society doesn't shun enough people |
| 20:36.20 | bmoraca_work | raden_work: that information does not get transmitted over SIP. |
| 20:36.25 | Naikrovek | right |
| 20:36.32 | Skeeter- | Naikrovek, would you mind uploading one of your custom rigntone |
| 20:36.41 | raden_work | bmoraca_work, but his panasonic phone system did it ( arguement im fighting :( ) |
| 20:36.41 | Qwell | Skeeter-: why do you ask? |
| 20:36.52 | beek | Skeeter-: bklang answered you in #vuc |
| 20:36.53 | Naikrovek | not a sip thing, but if you write it in your dialplan you can have the phone system IM your boss with jabber whenever anyone does anything |
| 20:36.53 | bmoraca_work | raden_work: so set up iSymphony and let him play with that |
| 20:36.55 | Skeeter- | rare1980_, thinks he got frauded |
| 20:37.32 | Naikrovek | Skeeter-: PM me your email address |
| 20:37.48 | raden_work | bmoraca_work, dont give him any ideas he would like that |
| 20:38.28 | bmoraca_work | raden_work: iSymphony is an application that will pretty much tell him what he wants and let him see who's on the phone with whom. it might get him off your back |
| 20:38.32 | Naikrovek | there's an app called AST that he might find useful if you can keep him out of the admin functions like disconnecting calls and pranking |
| 20:41.13 | *** join/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 20:42.20 | *** part/#asterisk dan__t (vpn@vpn.withparity.net) |
| 20:42.29 | *** join/#asterisk orn (~orn@194-144-94-105.du.xdsl.is) |
| 20:44.28 | orn | anyone have any idea why a 484 response would make asterisk send an ACK but not continue in the dialplan? |
| 20:44.32 | orn | http://pastebin.com/i7BahRLy |
| 20:44.52 | citywok | hmm, anybody ever gone to astricon? is it worthwhile? |
| 20:45.53 | mutax | WIMPy: HA- Using another PC with another kernel IT WORKSSS! !§"%!"§ my notebook is BROKEN: People, don't by lenovo T61's.... |
| 20:46.16 | citywok | mutax i have a pile of T61's that crash, have blown up completely, failed fans |
| 20:46.19 | citywok | pieces of _crap_ |
| 20:46.39 | citywok | personally i had a t61p and it didn't have too many issues |
| 20:46.43 | mutax | citywok: the speakers are lousy, and the performance is just... lowsy. |
| 20:47.47 | Corydon76-dig | citywok: depends upon whether you enjoy networking with other people |
| 20:47.50 | *** join/#asterisk CatLynx (sione@ocs.net) |
| 20:48.11 | citywok | that's the main goal actually -- see what other people have done and see what i could implement in my shop |
| 20:48.13 | *** join/#asterisk blaines (~blaines@64.16.22.201) |
| 20:48.24 | CatLynx | having issues with DAHDI |
| 20:48.39 | *** part/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 20:48.42 | Corydon76-dig | I will probably be there again this year, as will likely half of the development team or more |
| 20:49.31 | citywok | it seems like a good idea. in the last 9 months we dumped our entire old phone system and put together a (nearly) fully integrated * solution in our call cente |
| 20:49.36 | Corydon76-dig | We may even drive up to DC, which is where it will be this year |
| 20:49.57 | citywok | yea i saw that, i've never been to the other washington (i'm from seattle), so it'd be cool to spend a day doing that touristy crap |
| 20:50.15 | Corydon76-dig | Hah, a day... |
| 20:50.23 | CatLynx | I set the CID name to "Test Phone" and when the Analog line DAHDI/3-1 is set to exten number 6123 the name is display as "6123" not "Test Phone" when I change the exten to "14081236123" then it starts displaying "Test Phone" as the CID name. |
| 20:50.32 | Corydon76-dig | You could spend a month in DC and not see half of the good stuff |
| 20:50.49 | CatLynx | confused |
| 20:50.49 | hardwire | Corydon76-dig: I'll speak if you guys send me tickets :) |
| 20:50.53 | citywok | Yea i'm sure. i'd probably only spend a day or two though :) |
| 20:51.21 | Corydon76-dig | hardwire: if you speak, you get in free |
| 20:51.42 | hardwire | Corydon76-dig: I'll speak of the power of n.. and the extension that would never die. |
| 20:51.53 | Corydon76-dig | but that requires that your presentation be picked |
| 20:52.09 | hardwire | Corydon76-dig: oh.. it will be picked |
| 20:52.14 | hardwire | I radiate awesomeness |
| 20:52.18 | hardwire | ask file |
| 20:52.25 | Corydon76-dig | I'm considering driving up and staying at my parents' |
| 20:52.46 | hardwire | would totally take the ferry from Seward, AK |
| 20:53.25 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:55.09 | voxter | Hm. can any of you guys recommend a PSTN provider (SIP or PRI, SIP preferrably) that has the ability to a) turn up DIDs in every US state, b) provide an API to allow me to request and activate a new DID (or at least have their cust svc dept do it in under 48hrs) and c) not charge thru the nose? |
| 20:55.30 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 20:55.45 | smooth_penguin | callcentric maybe? |
| 20:55.55 | Chainsaw | voxter: Sure, pick any two. |
| 20:56.25 | CatLynx | Vitelity |
| 20:56.26 | voxter | chainsaw, i'm ok with it not being the cheapest, of course, i just cant meet level3's sip service commit of $20k/month |
| 20:56.48 | citywok | well i'd stay away from bandwidth.com |
| 20:56.59 | patrb | voxter, ask level3 to refer you to a reseller |
| 20:57.00 | citywok | flowroute has treated me pretty well, voicepulse was okay but slightly lower call quality |
| 20:57.15 | citywok | qwest is expensive for termination |
| 20:57.32 | citywok | xo wouldn't talk to me since i dont have any of their services |
| 20:57.39 | voxter | the 3rd, 4th level resellers like voicepulse wont do, i want to talk either directly to tier1, or one step away. |
| 20:57.51 | voxter | Vitelity customer service was reportedly not very good. |
| 20:57.57 | CatLynx | I like the fact FlowRoute displays who forwarded a call when it comes it, that works great for voice mail systems. |
| 20:57.59 | *** join/#asterisk JKac3BEq (~JKac3BEq@pl2.haspere.com) |
| 20:57.59 | voxter | I can ask level3 for a reseller reccommendation yes., |
| 20:58.16 | *** join/#asterisk resno (~bryan@cpe-098-026-005-130.nc.res.rr.com) |
| 20:58.20 | citywok | a level1 carrier isn't going to talk to you for $5k/mo |
| 20:58.23 | voxter | I'm going to need to scale this to thousands of users within a short amount of time, just not immediately. |
| 20:58.42 | *** part/#asterisk resno (~bryan@cpe-098-026-005-130.nc.res.rr.com) |
| 20:58.45 | citywok | then test it out on a smaller carrier thta will work with you now |
| 21:00.15 | CatLynx | Anyone konw why My Digium TDM410 is not displaying caller ID name when its CID number is set to 4 digits? but it will display when its set to a 11 digit number? |
| 21:00.47 | p3nguin | incoming calls? |
| 21:00.55 | CatLynx | outgoing |
| 21:01.10 | p3nguin | The telco is not accepting invalid lengths of CID. |
| 21:01.13 | Chainsaw | CatLynx: Telcos apply a filter. |
| 21:01.22 | CatLynx | this is internl not going to any proviers |
| 21:01.41 | Chainsaw | CatLynx: So you're providing CLI to a specific device? |
| 21:01.42 | p3nguin | erm |
| 21:01.42 | CatLynx | TDM410 FXS to SIP device |
| 21:02.00 | Chainsaw | CatLynx: Is it FXS or SIP? It can't be both. |
| 21:02.22 | p3nguin | The outgoing call originates from what device and terminates on what device? |
| 21:02.38 | thehar | what is the default naming scheme for a recorded queue call if it's never set? Just the UNIQUEID.format? |
| 21:02.44 | CatLynx | heh I ment to say this is a inhouse test setup testing analog phone plug in to a TDM410 FXS port calling a SIP device register to the same asterisk |
| 21:03.32 | p3nguin | catlynx: The outgoing call originates from what device and terminates on what device? |
| 21:03.52 | CatLynx | TDM410 making a outgoing call to the SIP phone |
| 21:03.59 | thehar | russellb: can you answer my question |
| 21:04.19 | p3nguin | catlynx: How are you setting the caller id value? |
| 21:05.37 | CatLynx | I set this in "chan_dahdi.conf" in the group for the FXS port |
| 21:05.38 | CatLynx | fullname = Test Phone |
| 21:05.38 | CatLynx | cid_number = 6123 |
| 21:06.13 | CatLynx | I also try "callerid= Test Phone <6123>" |
| 21:06.20 | CatLynx | had the same affect |
| 21:07.00 | CatLynx | this is on Asterisk 1.6.2.6 |
| 21:08.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:08.56 | CatLynx | wonder if its a bug |
| 21:09.05 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 21:09.47 | CatLynx | oh and I am using DADHI 2.2.1.1 |
| 21:09.56 | CatLynx | err DAHDI |
| 21:14.38 | *** join/#asterisk Zizou (~zizou@190.37.18.195) |
| 21:14.58 | *** join/#asterisk Skeeter- (~william@c75.154.173-9.clta.globetrotter.net) |
| 21:15.32 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 21:16.42 | Zizou | hi, im an asterisk beginner, ive been reading and now i understand a lot on how asterisk work, i did some test without using GUI, now im trying asteriskNOW but when i create a SIP extension aparently it doesnt really edit the sip.conf file, so my softphone doesnt register... |
| 21:18.51 | CatLynx | Zizou: it problay added the entery in "users.conf" |
| 21:19.01 | Tim_Toady | Zizou it edits the sip_aditional.conf |
| 21:19.10 | CatLynx | Zizou: do you see a login failer? |
| 21:19.24 | Tim_Toady | to make sure connext to asterisk cli and try "sip show peers" |
| 21:19.29 | Tim_Toady | connect* |
| 21:19.36 | Zizou | Tim_Toady, CatLynx so, what else should i do to get a basic sip phone work? |
| 21:19.52 | Tim_Toady | ad the extension on the web intrface |
| 21:20.01 | Tim_Toady | and try to register the phone |
| 21:20.39 | Zizou | Tim_Toady, i all ready do that... and nothing |
| 21:20.55 | Tim_Toady | connect to the asterisk cli |
| 21:20.59 | Tim_Toady | asterisk -rvvv |
| 21:21.04 | Tim_Toady | and see there whats going on |
| 21:21.24 | Tim_Toady | try "sip show peers" and se if the extension you addes is listed there |
| 21:21.29 | Tim_Toady | added* |
| 21:21.39 | Zizou | Tim_Toady, the command line doesnt recognize any command :S |
| 21:21.44 | Tim_Toady | ? |
| 21:22.30 | CatLynx | do you get anything at all if you type "sip ?" |
| 21:22.48 | Zizou | Tim_Toady, no such command |
| 21:23.02 | CatLynx | chan_sip problay did not load up due to same error |
| 21:23.11 | CatLynx | config file error? |
| 21:23.14 | Tim_Toady | which command? asterisk -rvv in shell or sip show peers in asterisk cli? |
| 21:23.30 | CatLynx | err same = some |
| 21:23.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:24.05 | Zizou | Tim_Toady, and the console doenst debug anything when i try to register |
| 21:24.27 | CatLynx | Zizou: what do you see when you type "sip ?" |
| 21:24.35 | Tim_Toady | is chan_sip loaded? |
| 21:24.44 | Zizou | Tim_Toady, lol yes i know, i did some test with asterisk in debian, without GUI, so i have some basics |
| 21:27.36 | Zizou | CatLynx, wtf doesnt have ? |
| 21:27.40 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:27.49 | [TK]D-Fender | Zizou: type "help sip" and pastebin the complete attempt & its results |
| 21:27.51 | [TK]D-Fender | ~pb |
| 21:27.52 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:27.53 | [TK]D-Fender | ^^^^^^ |
| 21:28.13 | Zizou | no such command sip |
| 21:28.22 | [TK]D-Fender | Zizou: that is not what I asked. |
| 21:28.35 | CatLynx | Zizou: ok this shows that the module chan_sip did not load |
| 21:29.08 | CatLynx | Zizou: type "module load chan_sip.so" and look at the messages on the CLI to see why it did not load up. |
| 21:29.11 | Qwell | Click the big orange button and restart. |
| 21:29.18 | Tim_Toady | Zizou chan_sip.so is not loaded or thers some prob with the configs, keel in mind that when you use a web gui is not a good idea to edit confs by hand |
| 21:29.47 | Zizou | CatLynx, ok |
| 21:29.49 | Zizou | Tim_Toady, |
| 21:29.50 | Zizou | ok |
| 21:30.25 | Zizou | CatLynx, Loaded chan_sip.so => (Session Initiation Protocol (SIP)) |
| 21:30.30 | Zizou | CatLynx, loaded |
| 21:30.42 | CatLynx | Zizou: see if your phone reg now |
| 21:30.51 | mutax | can somebody give me a hint what Received IE 29 not allowed (mt=5) means in lcr? google only gives references to the bible on that -.- |
| 21:30.57 | CatLynx | Zizou: sip show peers |
| 21:31.54 | *** join/#asterisk bakermd (~bakermd@38.104.0.102) |
| 21:32.05 | Zizou | CatLynx, yeah is monitorizing but doenst register, should i re-create the sip acount? |
| 21:32.21 | bakermd | Is there a way to, within a dialplan (say at hangup) get the call duration? |
| 21:32.55 | [TK]D-Fender | bakermd: "core show function CDR" |
| 21:33.25 | hardwire | voxter: yikes.. 20k? |
| 21:33.33 | voxter | hardwire: i believe. |
| 21:33.43 | hardwire | now I know why we aren't one of their clients. |
| 21:33.48 | hardwire | I was wondering about that earlier today |
| 21:33.59 | hardwire | I'm using etollfree for toll free atm, they use L3 |
| 21:33.59 | CatLynx | Zizou: the cli should show why its not registering |
| 21:34.02 | Zizou | CatLynx, wait let me try again, i didn reload |
| 21:34.56 | bakermd | [TK]D-Fender: Thanks - I see there that I can set things going into the CDR, but will it allow me to set a variable to one of its values, i.e. |
| 21:35.40 | Zizou | CatLynx, no, and doesnt show any debug |
| 21:35.59 | CatLynx | Zizou: you see the peer in "sip show peers" ? |
| 21:36.17 | Zizou | CatLynx, yeah |
| 21:36.17 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 21:37.32 | bakermd | SET(myvar=${CDR(duration)}) |
| 21:37.46 | doctorray | Does anyone recall which document references a button to the key number on polycom phones? |
| 21:37.58 | CatLynx | Zizou: i need to get going now, but look over your config make sure user name and pass are correct, make sure SIP phone is config right as well. |
| 21:39.20 | Zizou | CatLynx, yeah im looking that but ive been trying with x-lite and ekiga without luck |
| 21:41.34 | [TK]D-Fender | Zizou: "sip set debug on" <- look at the actual attempt |
| 21:42.30 | bakermd | [TK]D-Fender: As always, you are the man! Thanks for the help my friend :-) |
| 21:43.01 | [TK]D-Fender | bakermd: You're welcome |
| 21:49.19 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 21:49.32 | doctorray | If I place a call on hold to answer another call, is it possible for another extention to pick up my held call? |
| 21:49.56 | doctorray | or is parking the only way to accomplish that feature? |
| 21:52.35 | p3nguin | doctorray: Extension, no. Phone, yes. |
| 21:54.02 | doctorray | p3nguin: okay, s/extension/phone, both are polycoms. what feature should I be looking for? |
| 21:56.38 | p3nguin | ChannelRedirect() might be useful. |
| 21:56.40 | ChannelZ | SLA |
| 21:57.02 | *** join/#asterisk blaines (~blaines@75-171-125-14.phnx.qwest.net) |
| 21:57.12 | bmoraca_work | what would be causing Asterisk to be rejecting a REGISTER with a 401 Unauthorized and not prompt the phone to retry with credentials? |
| 21:57.14 | [TK]D-Fender | ChannelZ: LOL.... Magic Pink Unicorns do not come included with the software.... |
| 21:57.33 | ChannelZ | One can dream |
| 21:57.34 | [TK]D-Fender | bmoraca_work: its the phone that doesn't want to retry with credentials |
| 21:57.46 | [TK]D-Fender | ChannelZ: In TechniColor no less... |
| 21:57.47 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 21:57.51 | hardwire | bmoraca_work: responding with a 401 = send me creds or die |
| 21:58.52 | doctorray | p3nguin: I'll look into that, thanks |
| 21:59.03 | bmoraca_work | i understand that, but nothing in my configuration has changed and now two out of six phones don't send the credentials. of course, i don't seem to be getting an ACK either, so i wonder if there's something else at work |
| 21:59.06 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:59.53 | *** join/#asterisk blaines (~blaines@75-171-125-14.phnx.qwest.net) |
| 22:01.28 | ruben23 | hi is latency dependent to the capacity of your bandwidth..which voip also depend..? |
| 22:01.52 | bmoraca_work | latency is mostly dependent on prioritization delay and the clock speed of the line |
| 22:02.01 | bmoraca_work | neither factor is really under your control |
| 22:02.12 | doctorray | can a dialplan context be called when an extension places a call on hold? |
| 22:04.10 | *** join/#asterisk blaines (~blaines@75-171-125-14.phnx.qwest.net) |
| 22:04.37 | ChannelZ | I don't think ChannelRedirect will do you much good, easily anyway |
| 22:04.45 | ruben23 | bmoraca_work: even i have bigger bandwisth capacity, i stil can have latency issues, and voice issue with voip..? |
| 22:04.54 | bmoraca_work | yes |
| 22:05.49 | [TK]D-Fender | doctorray: You could probably have an app monitor the event via AMI. |
| 22:07.00 | ruben23 | bmoraca_work: what might a solution for it..? |
| 22:07.07 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 22:07.27 | bmoraca_work | ruben23: a closer ITSP or a different ISP that takes a better route to your current one |
| 22:07.37 | doctorray | [TK]D-Fender: hmm, good idea... I will look into that |
| 22:07.42 | mutax | Is there a lcr-expert awake? |
| 22:08.14 | mutax | I got outgoing calls somehow managed, only incoming are not working. lcr just hangs up while asterisk is trying to answer the line |
| 22:08.31 | mutax | or to be more specific, lcr does not seem to get on-hook |
| 22:09.52 | *** part/#asterisk Zizou (~zizou@190.37.18.195) |
| 22:10.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:12.51 | mutax | WIMPy: http://pastebin.com/N74sFbRr <- do you have any idea? |
| 22:14.14 | ruben23 | bmoraca_work: have you used voip..? |
| 22:16.03 | bmoraca_work | ruben23: i RUN an ITSP, so yes. i'm also intimately familiar with issues associated with VoIP |
| 22:16.13 | ChannelZ | hmm interesting I just noticed if you hit * while on hold, it stops the music for a fraction of a second while you're beeping |
| 22:17.15 | ruben23 | bmoraca_work: ow nice. whats your company-maybe i can test with you. im finding a good one. |
| 22:17.36 | *** join/#asterisk lbarth (~lbarth@62.216.165.71) |
| 22:17.37 | bmoraca_work | ruben23: i really only do business in California |
| 22:17.39 | lbarth | hello |
| 22:17.52 | bmoraca_work | and my biggest customer is myself (hosted PBXes are my main product) |
| 22:18.52 | ruben23 | bmoraca_work: im on the philippines, do i have chance for better voip connectivity.. |
| 22:18.56 | lbarth | I'm again working on an t38 udptl problem. http://pastebin.com/QCkt8g3U |
| 22:19.14 | lbarth | anyone has an hint how to debug that down? |
| 22:19.25 | citywok | ruben23: we use an MPLS network and backhaul our audi ofrom the philippines to Washington State |
| 22:19.48 | lbarth | i already tied different values for MaxDatagram but without any success |
| 22:19.54 | citywok | then terminate it in the US, 99.9% of our calls are to the U.S. so it's the cheapest way |
| 22:20.13 | ruben23 | citywok: hey, whats yor company..? |
| 22:20.25 | citywok | www.csgchannels.com |
| 22:21.10 | ruben23 | <PROTECTED> |
| 22:21.18 | citywok | for what? |
| 22:21.34 | ruben23 | for termination.. |
| 22:21.39 | voxter | citywok: can you tell me why you suggested to stay away from bandwidth.com for trunking? |
| 22:21.57 | citywok | ruben23: this is private, we don't do terminatino for anybody else. |
| 22:22.41 | citywok | voxter: i dislike them. they have the worst billing department i've ever seen (and for a telecom company, that's saying a lot), they have handed us off to a couple reps, and the newest one takes 24 hours to respond to anything at best |
| 22:23.08 | citywok | to be fair they have high quality calls, fantastic reliability, and if you have enough volume they are also pretty cheap |
| 22:23.21 | *** join/#asterisk Faithful (~Faithful@119.12.29.20) |
| 22:23.27 | citywok | in the last 18 months with them i haven't had a single outage |
| 22:23.45 | *** join/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 22:23.53 | ruben23 | citywok: CSG is a voip carrier right..? |
| 22:24.01 | citywok | ruben23: no, we are a call center |
| 22:24.12 | citywok | i was just explaining how we backhaul our own phone calls :P |
| 22:24.49 | ruben23 | <PROTECTED> |
| 22:25.15 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 22:26.42 | R-Guy | Anyone that can recommend or know-of a SIP Access Gateway which can support 500 or more 2-wire lines and loops lengths of up to 5km? |
| 22:27.23 | citywok | converting 500 analog lines in to sip? that sounds fun |
| 22:27.48 | citywok | you're probably going to need to use banks of FXS cards that do maybe 24 channels per card |
| 22:27.59 | R-Guy | To replace a number of small DMS switches, but using exisiting outside plant. |
| 22:28.05 | citywok | thats the only way i know of, though there may be a beter way at your scale |
| 22:28.26 | citywok | not cat5 pass-thru w/ sip phones? |
| 22:29.16 | R-Guy | The phones are 2w devices. |
| 22:29.51 | citywok | sounds like you need 24 port FXS cards then |
| 22:30.12 | citywok | and server(s) with a lot of PCI slots to hold them |
| 22:30.15 | [TK]D-Fender | .... |
| 22:30.17 | [TK]D-Fender | NO |
| 22:30.36 | R-Guy | I don't think they will support 5km loops and wouldn't have sufficient lightning protection. |
| 22:30.52 | citywok | oh shit, 5km? i read that as 5m. lol |
| 22:31.06 | [TK]D-Fender | First Digium recommends no more than TWO telephony interface cards per server, plus the fact that those cards are obscenely large and the server overhead costs is ridiculous |
| 22:31.07 | citywok | <-- no ideas at all then |
| 22:31.24 | Chainsaw | Patton makes gateways with up to 32 FXS ports. |
| 22:31.38 | Chainsaw | You may want to have a browse there. |
| 22:31.44 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 22:32.19 | citywok | [TK]D-Fender: we had a pair of 24 port analog cards and had a lot of crosstalk between them. and yea those things are retardedly big/expensive |
| 22:32.31 | citywok | thankfully they now occupy shelfspace |
| 22:32.49 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:41.19 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:44.57 | blaines | Anyone know why cisco 7940 phones will keep ringing after a call is answered? |
| 22:45.14 | p3nguin | I guess 'cause it's broken. |
| 22:45.34 | p3nguin | I've never had the phone keep ringing after answering it. |
| 22:46.18 | blaines | yea it's so strange |
| 22:46.38 | blaines | the asterisk server is co-located |
| 22:47.09 | blaines | not sure if the client's/datacenter's connection is failing |
| 22:47.28 | blaines | or if the server is just screwed |
| 22:47.52 | p3nguin | Got a sip debug? |
| 22:51.44 | Naikrovek | R-Guy: channel bank |
| 22:51.47 | Naikrovek | i think... |
| 22:52.31 | Naikrovek | 5km... interesting |
| 22:52.37 | Naikrovek | how many sites |
| 22:53.05 | Naikrovek | how many different places will/are the phones be/at |
| 22:53.44 | Naikrovek | i ask because this might be a good time to redesign the infrastructure a bit, if you're so inclined |
| 22:56.30 | blaines | p3nguin: unfortunately I don't have access to the Asterisk server at this time. Is there a way to definitively diagnose whether its a client or server problem? |
| 22:56.38 | blaines | err... without access to the server |
| 22:59.33 | p3nguin | I wouldn't think so. |
| 23:03.15 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 23:06.31 | citywok | well, if another sip client works great i'd say it would tell you it was the cisco phone |
| 23:06.45 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:06.53 | citywok | or at least help you identify the issue between the cisco phone itself and the server |
| 23:13.19 | *** join/#asterisk Faithful (~Faithful@121.91.118.214) |
| 23:25.12 | *** join/#asterisk Faithful (~Faithful@121.91.118.214) |
| 23:25.47 | R-Guy | Naikrovek: I am replacing an existing DMS switch. Phones are 2w and connected by outdoor copper. |
| 23:26.14 | Naikrovek | R-Guy: wow |
| 23:26.37 | R-Guy | Channel bank usually means TDM -- I don't want that. |
| 23:26.45 | Naikrovek | R-Guy: spread all over the place or in just a few places |
| 23:27.22 | R-Guy | Rural/suburban communities. |
| 23:27.29 | Naikrovek | are you a telephone company |
| 23:27.34 | Naikrovek | or part of one |
| 23:27.44 | R-Guy | I am working with a telco. |
| 23:27.52 | Naikrovek | this is very interesting to me |
| 23:27.56 | Naikrovek | sorry for all the questions |
| 23:27.59 | R-Guy | np |
| 23:28.19 | Naikrovek | mormons do this sometimes; they virtually buy a town and give everyone phone service |
| 23:28.24 | Naikrovek | at least i heard that |
| 23:28.38 | Naikrovek | buy all the houses, let church members live in them, etc |
| 23:28.39 | Naikrovek | anyway |
| 23:28.48 | Naikrovek | you're probably looking at telco gear here |
| 23:28.58 | Naikrovek | which i think very few of us, if any, know much about |
| 23:29.06 | R-Guy | The telco wants to dump the existing switches because of the cost of mantenance. |
| 23:29.11 | Naikrovek | yeah |
| 23:29.14 | Naikrovek | i can imagine |
| 23:29.48 | Naikrovek | why is TDM bad |
| 23:29.52 | Naikrovek | for this application |
| 23:30.31 | R-Guy | I have an IP switch -- TDM doesn't seem like a good fit. |
| 23:31.03 | Naikrovek | 500 phones, huh. man this is neat thinking about |
| 23:31.48 | Naikrovek | well the TDM doesn't extend into the public internet, that's only between the channel bank and the phone server |
| 23:32.01 | Naikrovek | once data leaves the phone server destined to the rest of the world, the tdm is gone |
| 23:32.09 | Naikrovek | it's all udp streams at that point |
| 23:34.05 | Naikrovek | there are definitely telco solutions for this, but i don't know what they are |
| 23:34.12 | R-Guy | For calls between subscribers, the media shouldn't even leave the access gateway (except for maybe lawful intercept) |
| 23:34.19 | Naikrovek | i have a friend that does, if you want me to ask, but i think the telco you're working with would know as well |
| 23:34.59 | Naikrovek | R-Guy: yeah asterisk can do that. asterisk can intercept if necessary and calls between channels don't leave the box, with analog lines |
| 23:35.21 | R-Guy | The telco knows the Nortel equipment -- they are asking us forthe new solution. |
| 23:35.37 | Naikrovek | voice goes from phone, to copper, to channel bank (in this scenario) to asterisk, back to channel bank, down copper to the other participant and then their handset |
| 23:35.54 | Naikrovek | you an record all, some, or none of the calls via asterisk |
| 23:36.27 | Naikrovek | the channel bank may be smart enough to bridge the lines internally in some cases, but asterisk can stay in the loop if you want it to |
| 23:36.46 | Naikrovek | lawful intercept would be an example of "if you want it to" |
| 23:39.16 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:40.03 | paulc | I helped some guy out with some Zyxel equipment a while back.. you put the box in the cabinet at the end of the street.. it did POTS service + ASDL (DSLAM) but POTS was fed via VoIP so you only had to connect the box to ethernet |
| 23:40.08 | paulc | it was pretty groovy to play with |
| 23:40.37 | paulc | would that kind of setup work? It means moving the equipment out of the CO and closer to the end cab I guess.. an option though, maybe.. |
| 23:47.32 | Naikrovek | paulc: he said that some endpoints were 5km from where the existing DMS switch is |
| 23:47.35 | p3nguin | Anyone know what would be the cheapest way to create a presence (virtual office) in another state which provides a mailing address? Mail forwarding services are expensive, and little or no mail may actually need to be sent to the virtual office. $500 per year for a service that might not ever be used is a bit extreme. |
| 23:48.32 | Naikrovek | p3nguin: hire someone a friend or relative, pay based on mail volume |
| 23:48.39 | Naikrovek | or something i dunno, really |
| 23:51.23 | p3nguin | I wish places would offer piece rate instead of a semi-annual or annual subscription. |
| 23:52.34 | Naikrovek | yeah |
| 23:52.35 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 23:52.44 | Naikrovek | they can't earn money for nothing with piece rates though |
| 23:52.46 | *** join/#asterisk fogrider (~fog@99-202-105-9.pools.spcsdns.net) |
| 23:53.00 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 23:53.13 | R-Guy | paulc: I need to connect to the cable pairs at the central office -- no facility to connect at the curb at this time. |
| 23:53.37 | joobie | hey guys, is there a way to receieve an SMS into asteirsk? im trying to seutp something where a user can SMS a number and then it reach asterisk.. not sure of an autoamted way to do this.. |
| 23:54.18 | Naikrovek | joobie: don't think so. US carriers (you may not be in the US) don't do SMS like that. SMS here is wireless only, except for some websites |
| 23:54.22 | Naikrovek | and those are send only |
| 23:54.29 | Naikrovek | oh duh you're in .au |
| 23:54.33 | Naikrovek | nevermind me |
| 23:54.39 | fogrider | I got an issue with an asterisknow machine, was working fine. then power went out causing a reboot. System now will not let sip work. i can get asterisk to run but sip will not reboot, just hangs. |
| 23:55.02 | Naikrovek | joobie: i don't know asterisk's SMS capabilities but i know they're few. manxpower knows more about this |
| 23:55.15 | Naikrovek | sip is not a process |
| 23:55.17 | Naikrovek | also |
| 23:55.21 | Naikrovek | go to #asterisknow |
| 23:55.30 | Naikrovek | we don't know that system anywhere near well enough to support it here |
| 23:56.02 | Naikrovek | sip is a protocol, they don't boot, reboot, or hang. |
| 23:56.47 | joobie | damn |
| 23:57.02 | fogrider | ok... well when i log into the portal for asterisk, it takes about 5 minutes until sip functions are avaiable. and if i run the command, sip reload, it doesn't reload. |
| 23:57.03 | joobie | sounds like it might be a bit hard to achieve |
| 23:57.11 | joobie | i was reading about all these sms functionality inbuilt into asterisk |
| 23:57.15 | joobie | was hoping there was a native way to do this |
| 23:57.24 | Naikrovek | joobie: there's some SMS documentation in the source code, beyond that i personally can't help |
| 23:57.27 | Naikrovek | manxpower knows more |
| 23:57.30 | Naikrovek | wait for him to show up |
| 23:57.52 | Naikrovek | fogrider: you're going to need to show us a log of it before we can do much |
| 23:58.03 | paulc | joobie: Do you have SMS over analog phone lines there? I know they do in the UK, wasn't sure about Australia |
| 23:58.04 | Naikrovek | perhaps sip debug output |
| 23:58.42 | fogrider | Naikrovek: would u be able to help if i show you the output i get? |
| 23:58.52 | Naikrovek | fogrider: if not me, someone else |
| 23:59.12 | Naikrovek | i'm not a very good asterisk diagnosis person yet but there are many in here who are |
| 23:59.19 | Naikrovek | and don't look now, but they're reading this |
| 23:59.23 | Naikrovek | and paying attention |
| 23:59.39 | Naikrovek | so state your problem, and wait as long as your patience will allow |
| 23:59.41 | Naikrovek | someone will respond |
| 23:59.44 | fogrider | i forgot to say, when i first rebooted i got a asterisk message about "does the asterisk.cfg exist?" so i moved that file and rebooted, then asterisk would run but now sip wouldn't work |
| 23:59.45 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |