| 00:26.05 | *** part/#asterisk kharwell (kharwell@nat/digium/x-adhfwctqzjicnesz) |
| 00:39.59 | Get_The_Fish | did we just remove pjsip reload from 13.7??? |
| 00:47.05 | WIMPy | Hmmm |
| 00:47.28 | WIMPy | wonders if it's possible to find out if a variable is unset or empty. |
| 00:48.23 | MaliutaLap | WIMPy: in what language? |
| 00:48.29 | WIMPy | Dialplan |
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| 00:53.43 | MaliutaLap | WIMPy: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_EXISTS |
| 00:53.49 | djiboutiii | Anyone pretty familiar with rtp packets? |
| 00:54.13 | djiboutiii | When I reach my asterisk server by phone, I can see these RTP packets Sent RTP packet to 95.211.119.239:64374 (type 00, seq 063708, ts 092320, len 000160) |
| 00:54.32 | djiboutiii | However, once I forward it to a cell and answer it. I see RTP packets missing the SEQ and TS information |
| 00:54.59 | djiboutiii | Sent RTP P2P packet to 152.184.3.208:25066 (type 00, len 000160) |
| 00:55.02 | djiboutiii | like that ^^ |
| 00:55.16 | djiboutiii | Any idea why? Hoping there's a clue there as to why I can't get audio on forwarded calls |
| 00:55.28 | WIMPy | That description is VERY vague. But probably it does somethin else than ISNULL. Let's make a test |
| 00:57.55 | WIMPy | Nope. Returns 0 in both cases. |
| 00:58.01 | MaliutaLap | WIMPy: can't say I have seen an ISNULL or ISSET |
| 00:59.15 | WIMPy | thinks the "(data)" means it won't make a difference. |
| 00:59.33 | WIMPy | I don't have an ISSET. Where's that hidden? |
| 01:00.07 | WIMPy | Not on the wiki, either. |
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| 01:12.17 | MaliutaLap | I haven't seen one, not for dialplan |
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| 08:44.49 | wyoung | werd |
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| 11:05.57 | feezy | Hi all! I have a little problem on a fresh install of asterisk, with very basic configuration, my "extensions.conf" file doesn't seems to be loaded... I have 2 ppers connected, but when i want to call from one to an other, i have thhis error in the console: " Call from 'luffy' (192.168.1.50:5060) to extension '11' rejected because extension not found in context 'my_context'". If someone has an idea ? :) Thanks in advance! |
| 11:05.57 | feezy | * I'm on a 13.7.0 version |
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| 12:56.59 | file | la la la |
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| 13:08.49 | [TK]D-Fender | feezy: You'll have to actually show us |
| 13:08.57 | [TK]D-Fender | feezy: "dialplan reload" from CLI |
| 13:09.17 | [TK]D-Fender | feezy: "ls -la /etc/asterisk" |
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| 13:09.39 | [TK]D-Fender | feezy: "cat /etc/asterisk/extensions.conf" |
| 13:09.41 | [TK]D-Fender | ~pb |
| 13:09.41 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 13:09.43 | [TK]D-Fender | ^^^^^ |
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| 13:28.49 | [TK]D-Fender | [08:08][TK]D-Fenderfeezy: You'll have to actually show us |
| 13:28.50 | [TK]D-Fender | [08:08][TK]D-Fenderfeezy: "dialplan reload" from CLI |
| 13:28.52 | [TK]D-Fender | [08:09][TK]D-Fenderfeezy: "ls -la /etc/asterisk" |
| 13:28.53 | [TK]D-Fender | [08:09][TK]D-Fenderfeezy: "cat /etc/asterisk/extensions.conf" |
| 13:28.55 | [TK]D-Fender | ~pb |
| 13:28.55 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 13:28.57 | [TK]D-Fender | ^^^ |
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| 13:47.50 | feezy | Nobody to help me ? :/ |
| 13:53.07 | [TK]D-Fender | I answered twice |
| 13:53.17 | [TK]D-Fender | And you keep quitting |
| 13:55.00 | feezy | Sorry... I had connexion problem :/ |
| 13:55.56 | [TK]D-Fender | [08:28][TK]D-Fender[08:08][TK]D-Fenderfeezy: You'll have to actually show us |
| 13:55.57 | [TK]D-Fender | [08:28][TK]D-Fender[08:08][TK]D-Fenderfeezy: "dialplan reload" from CLI |
| 13:55.59 | [TK]D-Fender | [08:28][TK]D-Fender[08:09][TK]D-Fenderfeezy: "ls -la /etc/asterisk" |
| 13:56.01 | [TK]D-Fender | [08:28][TK]D-Fender[08:09][TK]D-Fenderfeezy: "cat /etc/asterisk/extensions.conf" |
| 13:56.02 | [TK]D-Fender | ~pb |
| 13:56.02 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 13:56.04 | [TK]D-Fender | ^^^ |
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| 14:09.01 | feezy | Thanks a lot!! I find out wat was my problem: Since i'm starting from scrash to learn how it works, i didn't loaded the 'pbx_config.so' module. But now, i have an other problem: All is in the Pastbin: http://pastebin.com/R4VsF3U0 |
| 14:10.45 | Gugge | seems like it cant find kali as a peer, and uses it as a hostname |
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| 14:17.40 | [TK]D-Fender | feezy: you have [11] , [12] , etc |
| 14:17.55 | [TK]D-Fender | feezy: you are dialing a SECTION ENTRY, not a "username" |
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| 14:22.04 | feezy | ok, i just modify my sip.conf and change [11] whith [kali] and same for the other, and now: RINGING!! |
| 14:22.19 | feezy | Bug error when answer lol, i get closer! |
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| 15:14.24 | onebree | Good day, everyone. |
| 15:15.23 | onebree | What is the PJSIP equivalent of: "sip set debug peer <name>" |
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| 15:18.07 | [TK]D-Fender | pjsip <tab> |
| 15:18.48 | onebree | [TK]D-Fender: thank you. |
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| 15:22.41 | onebree | Me and another dev are working on a remote asterisk server. Once we enter "asterisk -r", the colors of the console change. Is there a way to set them to a different scheme? |
| 15:27.21 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+CLI+Configuration |
| 15:27.33 | [TK]D-Fender | +c |
| 15:38.02 | onebree | What is +c? |
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| 15:40.04 | [TK]D-Fender | asterisk -rvvvvvvvvvvvvvc |
| 15:41.17 | onebree | Oh, okay, thank you [TK]D-Fender |
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| 15:51.30 | ertyuio | hi |
| 15:51.56 | ertyuio | i was a time working with asterisk |
| 15:52.07 | ertyuio | now video fetaure is that possible on asterisk ? |
| 15:52.55 | [TK]D-Fender | yrou wording is very broken. |
| 15:53.03 | [TK]D-Fender | People have been using video with * for years |
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| 17:03.47 | ertyuio | since when ? |
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| 17:07.19 | [TK]D-Fender | YEARS |
| 17:07.24 | [TK]D-Fender | Possibly a decade |
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| 17:07.34 | [TK]D-Fender | maybe more. |
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| 17:14.19 | shido6 | history |
| 17:14.40 | [TK]D-Fender | ancient |
| 17:14.45 | shido6 | bows |
| 17:14.52 | shido6 | am I? |
| 17:14.54 | [TK]D-Fender | pwns these games.... |
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| 18:26.07 | onebree | test message |
| 18:26.16 | onebree | Great, Kiwi has not bugged on me |
| 18:26.24 | [TK]D-Fender | grades it an "F" |
| 18:40.21 | Get_The_Fish | Anyone here have polycom MWI working with PJSIP? |
| 18:44.16 | Get_The_Fish | Is everyone still using chan_sip? |
| 18:44.32 | WIMPy | No |
| 18:45.00 | onebree | Get_The_Fish: We are using chan_sip in production, but testing out pjsip. |
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| 19:06.49 | qakhan | hi all, |
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| 19:07.38 | qakhan | which .ext is good quality and low space of Monitor() |
| 19:07.46 | qakhan | gsm or wav? |
| 19:07.57 | [TK]D-Fender | How big are each of them? |
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| 19:08.12 | [TK]D-Fender | How do they sound to you? |
| 19:08.24 | qakhan | i did not try wav |
| 19:08.36 | qakhan | gsm is ok but not good |
| 19:10.11 | [TK]D-Fender | Well go try |
| 19:10.25 | [TK]D-Fender | It would have taken less time than asking. |
| 19:11.17 | qakhan | but it is also better to take some suggestion from guru like you ;) |
| 19:11.25 | [TK]D-Fender | Use your own ears |
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| 19:12.19 | qakhan | oka |
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| 19:17.21 | onebree | s/oka/okra |
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| 19:55.49 | wyoung | s/onebree/what are you on about?/ |
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| 21:46.49 | grim_ | anyone have cisco 7965 connecting to asterisk via tftp? I have a bunch of sccp that I'm converting to sip, but they are failing the validation of the config SEPXXXXXXXXXXXX.cnf.xml |
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| 22:23.56 | *** join/#asterisk Phlunk3 (~Phlunk3@mail.tw.co.nz) |
| 22:25.33 | Phlunk3 | I have a small issue, hope someone can steer me in the right direction to find a solution... basically when my asterisk server receives a call, it passes it on to my sip client however the calleridnumber is incorrect (it is the number being called) although calleridname has the correct number. Any suggestions as to where I should start looking? |
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| 22:30.39 | [TK]D-Fender | Phlunk3, Show us the complete call with SIP DEBUG enabled |
| 22:30.41 | [TK]D-Fender | ~pb |
| 22:30.42 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 22:30.42 | [TK]D-Fender | ^^^ |
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| 22:33.55 | Phlunk3 | bit new, this would be done by enabling some debug level in the cli tool and then just watching the output in cli while the call is made? |
| 22:34.16 | [TK]D-Fender | SIP DEBUG and verbose 10 |
| 22:37.13 | Phlunk3 | http://paste.lisp.org/display/306403 |
| 22:37.36 | Phlunk3 | hope no passwords in there lol. |
| 22:39.04 | [TK]D-Fender | From: "6475711060" <sip:075711060@27.111.14.65>;tag=as0f30ef2d |
| 22:39.05 | [TK]D-Fender | To: <sip:072811176@192.168.222.206:5060> |
| 22:39.14 | [TK]D-Fender | well we see the call come in from that CID, to that # |
| 22:39.38 | [TK]D-Fender | And your OUTGOING call has the SAME #'sFrom: "6475711060" <sip:072811176@192.168.222.206>;tag=as6f0a189e |
| 22:39.55 | [TK]D-Fender | So * is not doing anything to this at all |
| 22:40.26 | Phlunk3 | thanks, must be the clients then I guess. |
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| 22:41.41 | Phlunk3 | basically all clients see calleridnumber as the 072811176 and the calleridname as 647571060. I am just very new to this system so not even sure if asterisk was responsible for assigning those vars or if they're being generated in clients :/ |
| 22:43.06 | *** join/#asterisk slav3_sergal (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 22:43.24 | [TK]D-Fender | Came in looking like that, went out loking the same |
| 22:44.05 | Phlunk3 | tx |
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| 22:44.22 | [TK]D-Fender | Easy call at least |
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| 22:51.23 | mcargile | Trying to do a WebRTC phone using Asterisk 11 and Chrome keeps complaining about the security on my wss websocket. I have tlsenable=yes in http.conf and valid keys. |
| 22:52.51 | mcargile | Firefox has no issue |
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