| 17:46.44 | *** join/#asterisk infobot (ibot@rikers.org) |
| 17:46.44 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 17:48.26 | DanQuinney | sorry, that was a typo, i set `maximum_expiratio` |
| 17:48.30 | DanQuinney | *maximum_expiration |
| 17:48.48 | salviadud | I'm not sure if my fxo signalling is correct, I selected north america, but I am in mexico |
| 17:49.04 | salviadud | Where can I find a resource that tells me what to use? |
| 17:49.37 | salviadud | Well, it should be fxs signalling, since I have fxo ports... |
| 17:52.17 | salviadud | Ok, what I'm getting at is that I might not have the proper config for tones, such as hangup |
| 17:56.46 | salviadud | oh, I just found out I don't have an indications.conf ... |
| 17:58.17 | salviadud | Em, what module do I reload so indications.conf takes effect? |
| 17:58.24 | *** join/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 17:59.22 | salviadud | I just restarted everything... |
| 18:03.20 | salviadud | works fine, feel so fresh |
| 18:03.27 | *** join/#asterisk Pegasus_RPG (~Thunderbi@47.142.205.225) |
| 18:07.14 | Pegasus_RPG | Hello. I'm enabling direct SIP calling on my * box (v11.13 ATM) and UDP calls work fine. But when I try using TLS, * demands authentication. What should I be looking for in my configuration? |
| 18:16.37 | *** join/#asterisk overyander (~jeff@12.49.160.131) |
| 18:25.14 | wabbits | Pegasus_RPG you mean direct media? |
| 18:27.04 | *** join/#asterisk fridgefire (~michiel@unaffiliated/michielbrink) |
| 18:27.35 | Pegasus_RPG | wabbits: Where a standalone SIP phone can call sip:jdoe@pbx.example.com |
| 18:28.03 | Pegasus_RPG | without authenticating. (* will of course make sure that the domain is local and that jdoe exists.) |
| 18:28.29 | wabbits | so unless you are pbx.example.com you can only see the debug from your client |
| 18:28.41 | *** part/#asterisk voipmonk (~voipmonk@138.68.2.1) |
| 18:28.43 | Pegasus_RPG | I am |
| 18:28.44 | [TK]D-Fender | Show the call |
| 18:29.32 | wabbits | I am into federated voip and make calls like this all the time but I don't use tls. |
| 18:29.47 | Pegasus_RPG | [TK]D-Fender: the SIP trace? |
| 18:29.57 | [TK]D-Fender | The actual call attempt |
| 18:30.12 | [TK]D-Fender | SIP debug from * CLI, verbsoe 10 |
| 18:30.28 | wabbits | with sip trace inline |
| 18:34.46 | Pegasus_RPG | It boils down to * replying to a TLS connection attempt with SIP/2.0 401 Unauthorized |
| 18:35.28 | Pegasus_RPG | http://paste.debian.net/956199/ |
| 18:35.44 | [TK]D-Fender | Found peer 'HomePhone' for '0000000000' from [client public IP]:41580 |
| 18:35.56 | [TK]D-Fender | No, it comes down to the fact it's finding a MATCH to TRY to auth against |
| 18:36.06 | [TK]D-Fender | Your test is looking like a VALID PEER |
| 18:36.14 | Pegasus_RPG | ohhh |
| 18:36.48 | Pegasus_RPG | has a few SIP endpoints on the same network |
| 18:38.20 | Pegasus_RPG | But then why isn't this a problem using UDP? |
| 18:51.18 | [TK]D-Fender | what TRANSPORT is your peer set to? |
| 18:51.47 | [TK]D-Fender | that's why |
| 18:52.05 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 18:52.05 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 18:54.39 | *** join/#asterisk miralin (~Thunderbi@91.237.94.11) |
| 18:57.14 | Pegasus_RPG | [TK]D-Fender: transport=tls,udp |
| 18:57.34 | [TK]D-Fender | probably gets rtied to whatever registered against it <- |
| 18:57.47 | [TK]D-Fender | Which is the peer & port from the contact |
| 18:57.51 | *** join/#asterisk pavelsc (~Miranda@93.125.107.16) |
| 18:58.02 | [TK]D-Fender | Don't use the same softphone for your testing of this |
| 18:58.09 | [TK]D-Fender | that is a broken approach |
| 18:58.10 | Pegasus_RPG | Sure enough, if I change the peers to type=user, the test client works using TLS |
| 18:59.55 | Pegasus_RPG | sure |
| 19:00.08 | Pegasus_RPG | thanks for the direction |
| 19:01.04 | [TK]D-Fender | np |
| 19:20.58 | pavelsc | Hello guys. I looked through CEL docs and there everything is ok. CHAN_START event gives the information that callee is Bob in docs. But the reality is not so good. CHAN_START event gives me nothing. cid_name, cid_num, cid_ani, cid_dnid are empty. Exten is 's', and context is 'public'. Moreover <Bob's> channel always lacks information. Is it normal? Is there a common approach to parse cel data effectively? The code of my parser |
| 19:20.59 | pavelsc | looks like a hundred of tons of bad smelling conditional non-intuitive logic. For example when Alice calls to IVR and then selects queue with follow call user routing strategy, and agent Bob answers and then transfers Alice to Carrol then there is no chance to understand what is going on in CEL. |
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| 19:43.44 | *** join/#asterisk voipmonk (~voipmonk@138.68.2.1) |
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| 19:46.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 20:30.36 | Naikrovek | what the heck is up |
| 20:36.28 | *** part/#asterisk Pegasus_RPG (~Thunderbi@47.142.205.225) |
| 20:40.28 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 20:41.11 | salviadud | After fiddling with indications.conf so that I use values from my country, I'm dropping calls from Estados Unidos, that's not good. |
| 20:41.17 | salviadud | Some clients are from there |
| 20:41.38 | salviadud | So, I compared the values and "mixed" them up |
| 20:41.46 | salviadud | Am I on the right path here? |
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| 21:06.52 | Naikrovek | hmm. apparently the entire crew of people that were here when I lurked in this channel are gone. sadface.aiff |
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| 21:47.36 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:52.31 | salviadud | [TK]D-Fender, could you school me on dahdi? |
| 21:52.43 | salviadud | I recently played around with indications.conf |
| 21:53.01 | salviadud | So that I get the right settings for my country, but I'm dropping calls that originate from the US |
| 21:53.22 | salviadud | So, I added the values that exist on the US side that don't exist on MX |
| 21:53.27 | salviadud | I wonder if that'll solve it. |
| 22:26.45 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:27.22 | *** part/#asterisk SleepyCow (SleepyCow@pool-173-70-137-156.nwrknj.fios.verizon.net) |
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| 22:49.34 | salviadud | Well, I took a read and edited /etc/dahdi/system.conf |
| 22:49.50 | salviadud | indications.conf will be ignored, right? |
| 22:58.58 | [TK]D-Fender | no |
| 22:59.08 | [TK]D-Fender | indications goes BEYOND just DAHDI |
| 22:59.21 | [TK]D-Fender | and AFAIK is used by it for progress detection, etc |
| 22:59.25 | [TK]D-Fender | if you enabled that |
| 23:01.17 | salviadud | AFAIK? |
| 23:01.45 | salviadud | as far as you know, sorry |
| 23:02.32 | salviadud | So, after setting my loadzone correctly in /etc/dahdi/system.conf , is it worthwhile to set my indications.conf as well? |
| 23:03.04 | [TK]D-Fender | YES |
| 23:03.08 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:03.11 | [TK]D-Fender | Do the WSHOLE jib. |
| 23:03.18 | [TK]D-Fender | WHOLE JOB |
| 23:04.15 | salviadud | alright, got it |
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