| 00:07.49 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ppnyzflnbcwsolqw) |
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| 00:55.54 | hursjohn | I'm trying to build a recording and anouncement system for my switch using Asterisk. I have a E&M span comming in with a Digium T1 card. I have the framing all set up and I'm getting the call coming in. I set up the signalling as 'featdmf'. call is coming in, but I'm not seeing any ANI to route the call. turning debug on I see a bunch of DTMF coming in on the trunk. How would I go about |
| 00:55.54 | hursjohn | grabbing these DTMF into a variable in my context to route the call to an appropriate annoucement? https://pastebin.com/jSTMMZvA |
| 01:19.08 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.1 (2018/02/13), Standard: 15.2.1 (2018/02/13); 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 |
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| 01:22.24 | aissar | hi guys |
| 01:22.35 | kai[El] | hi |
| 01:22.49 | aissar | What's the difference between thats channels? |
| 01:22.51 | aissar | -- Executing [s@call-from-queue-connected:1] NoOp("Local/5213@from-queue-00000147;1", "") in new stack |
| 01:23.20 | aissar | -- Executing [s@quality-survey:1] NoOp("Local/5213@from-queue-00000147;2", "") in new stack |
| 01:23.57 | aissar | what means ;#number after the channel name? |
| 01:24.54 | aissar | I'm trying set a variable in call-from-queue-connected context to read in quality-survey... |
| 01:28.41 | aissar | but don't work like in SIP channels |
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| 01:36.17 | rmudgett | aissar: Local channels always come in pairs with an implicit bridge between them. The ;1 and ;2 denote the two channels of a pair. |
| 01:37.24 | aissar | thanks for the answer rmudgett |
| 01:37.54 | aissar | do you know some way to set a variable between that pairs? |
| 01:38.34 | rmudgett | https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance |
| 01:39.41 | aissar | I tried that way... but don't work for me |
| 01:40.01 | aissar | In SIP channels works like a charm |
| 01:41.26 | rmudgett | Inheritance only happens when the channel is created. The dialing channel can set variables before dialing that the dialed channel inherits. |
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| 01:44.39 | aissar | Ok |
| 01:46.13 | aissar | thanks rmudgett |
| 01:46.33 | aissar | good bye |
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| 02:05.00 | snadge | im just looking at a call trace to try and figure out whether t.38 actually worked or not |
| 02:05.23 | snadge | in the invite line i can see |
| 02:05.26 | snadge | a=cdsc:1 image udptl t38 |
| 02:05.41 | snadge | should the SIP 200 OK that comes back also include t38? |
| 02:06.25 | snadge | or should i just tell the customer.. the 1980s called, they want their technology back ;) |
| 02:07.48 | snadge | according to some random cisco documentation it does.. thanks guys :P |
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| 02:32.53 | kai[El] | bed time |
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| 14:52.12 | alfabit | I have multiple polycom ip 6000 conference phones attempting to register with asterisk. first one works as expected second does not. First one registers to 5060 and seems to tie up the port?! What am I missing any tips? |
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| 15:42.19 | igcewieling | alfabit: Servers can have the same port serve more than one client, like web servers do. |
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| 19:02.02 | jamesaxl | Hello |
| 19:08.36 | jamesaxl | this is my macro that I use for recording incoming calls https://bpaste.net/show/7fc9256d4607. My problems is : when I have transfer from operator to an other queue, the monitor inherit auto and the calls is recorded in the folder of the operator that did the transfer, so I can have the path of video file in data base but I can have any audio file in the last operator that took the call. Could you suggest |
| 19:08.38 | jamesaxl | for me a solution please |
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| 20:18.43 | kfife | How does the Asterisk SIP channel driver know the loss and jitter on the SEND side? Obvously only the remote agent knows if a UDP packet was lost, so with what frequency does remote SIP User Agent report that to the other side? |
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| 22:19.18 | chris349 | I have a sip trunk working great with no issues, but now I try to call another destiation on the trunk and the trunk gives an error about a digest mismatch. How can I force asterisk to authenticate based on user and password in sip.conf and not try to authentication based on the callerid number? |
| 22:24.48 | salviadud | chris349, could you do a sip debug so we can see the whole error? |
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| 22:27.54 | chris349 | Full error is: username mismatch, have <CALLERID>, digest has <USERNAME IN SIP.CONF> |
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| 22:31.26 | salviadud | Which version are you running? |
| 22:31.38 | chris349 | Asterisk 13.7.2 |
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