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| 01:24.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); 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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| 12:41.51 | vtadesh | Hi |
| 12:42.23 | vtadesh | can anybody help me to get the parent channel in attended transfer context? |
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| 13:24.31 | [TK]D-Fender | vtadesh, No way to get this |
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| 15:33.26 | florolf | hi |
| 15:34.30 | florolf | asterisk is telling me it's rejecting a call because the extension is not present in the dialplan/context, but it is: https://p.rrbone.net/paste/GLFwHTbR#gSBVVJ09l++0jkP8L8TNJchtivgaBycUdOl-zDeu0Yr |
| 15:34.34 | florolf | what am i missing? |
| 15:35.02 | florolf | googling this is pretty difficult, as it mostly turns up cases where people's dialplan didn't load for some reason or another |
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| 15:39.59 | igcewieling | show the *actual* dialplan. |
| 15:40.56 | [TK]D-Fender | the dump looks pretty solid. Do show the actual dialplan code, and enable SIP debug and place another call as well. |
| 15:43.01 | florolf | https://p.rrbone.net/paste/uQn2PAH1#ZL50pKmiM1Gg40grsZPC694SjxciCya2bOqjiLKjzLH |
| 15:45.46 | florolf | ah, the sip debugging output was helpful. the softphone i was using didn't honor the peer name i told it to use |
| 15:46.14 | florolf | though it did use that for registering *sigh* |
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| 15:53.11 | florolf | nevermind, the actual problem was that i used sip:1000 for dialing, but asterisk interpreted 1000 as the domain (and no extension within that domain, defaulting to s) |
| 15:54.03 | [TK]D-Fender | Upgrading from "looking" to "really looking" tends to turn up some interesting stuff most of the time... |
| 15:54.28 | florolf | not sure what the correct behavior is here, but i'm going to default to assuming PEBCAK |
| 15:54.29 | [TK]D-Fender | The basic verbose emssaging vs dialplan dump do look pretty convining though.. I understand the confusion |
| 15:54.35 | florolf | thanks for the hints! |
| 15:54.55 | [TK]D-Fender | you're welcome |
| 15:56.46 | igcewieling | it was a terrible error message |
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| 18:00.33 | tuxd00d | How do we keep Asterisk in the audio path? Dealing with a strick firewall. |
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| 18:08.53 | igcewieling | directmedia=no |
| 18:12.12 | tuxd00d | Thatâs what I thought. |
| 18:13.00 | tuxd00d | sip show peers indicates âDirectMedia : Noâ |
| 18:13.07 | tuxd00d | I mean sip show peer |
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| 19:46.21 | polysics | hey! good evening! |
| 19:46.33 | polysics | is there a way to set a SIP header on a "channel originate", please? |
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| 20:18.17 | [TK]D-Fender | use a local channel and set it before doing a Dial() yourself |
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| 20:54.14 | polysics | [TK]D-Fender: not from the CLI |
| 20:55.08 | [TK]D-Fender | depends what you are dialing in the end |
| 20:55.24 | [TK]D-Fender | And CLI originate is pointless anyway given the alternatives. |
| 20:55.42 | [TK]D-Fender | AMI or Call-files are far better. |
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| 21:05.33 | polysics | yup, I was just testing something |
| 21:05.42 | polysics | now I have another problem that is more puzzling |
| 21:06.00 | polysics | no matter what I say, I get "MACHINE" out of AMD |
| 21:07.59 | [TK]D-Fender | You're clearly a Cylon and should be put down like the toaster you are... |
| 21:21.26 | polysics | that is the only explanation I have. |
| 21:21.39 | polysics | like, I have tried saying literally anything |
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| 21:40.29 | polysics | I am really out of ideas as to why AMD is ALWAYS returning machine |
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