| 00:00.39 | Korolev | file, that makes sense, because that's all I find everywhere, functions |
| 00:00.55 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL |
| 00:02.00 | RovingWriter | beat me to it |
| 00:03.17 | Korolev | remote_uri, there we go! thank you! |
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| 00:11.06 | Korolev | man! two hours and five minutes later, I'm finally done. You guys rock! |
| 00:12.32 | fauxalliance | thats not a bad turnaround time, thanks for reading the manual! :) |
| 01:37.28 | elcontrastador | https://gist.github.com/elcontrastador/ce6081a9d2e1ca901b0ea6746bbda9eb |
| 01:38.08 | elcontrastador | getting make errors on 15.0.0rc1...any suggestions. looks like a dependency that configure script didnt pick up |
| 01:43.16 | fauxalliance | /usr/bin/ld: cannot find -lssl |
| 01:43.18 | fauxalliance | /usr/bin/ld: cannot find -lcrypto |
| 01:43.19 | fauxalliance | https://stackoverflow.com/questions/16710047/usr-bin-ld-cannot-find-lnameofthelibrary/32502403 |
| 01:44.51 | *** join/#asterisk x86- (~x86@ool-4570f1e8.dyn.optonline.net) |
| 01:48.56 | x86- | I'm getting an issue : chan_sip.c:23225 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:anonymous@sip.flowroute.com>;tag=as04e49904' |
| 01:49.19 | x86- | checked the files for anonymous .. I have it configured right. |
| 01:49.40 | fauxalliance | allow anonymous inbound sip in the sip settings? |
| 01:49.52 | fauxalliance | probably not |
| 01:52.52 | x86- | where would that be |
| 01:53.01 | x86- | I checked the config files searched for the word anonymous |
| 01:54.32 | fauxalliance | sip.conf allowguest=yes |
| 01:55.18 | fauxalliance | [23:23] <x86-> I checked the config files searched for the word anonymous >perhaps read the manual or google the issue |
| 01:58.44 | x86- | changed it and still getting the same error |
| 01:59.34 | fauxalliance | did something overwrite your configuration, like a GUI maybe |
| 02:00.51 | fauxalliance | and do you have an inboud route to catch such calls |
| 02:07.00 | fauxalliance | hmm, gsm gateway with chan_dongle... thats where chan_mobile went |
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| 07:14.16 | Guesty | How can Asterisk say "WARNING[11844][C-00004feb]: res_musiconhold.c:1438 local_ast_moh_start: Unknown format 'wav' -- defaulting to SLIN" When the file has the extension .GSM? It confuses me a lot |
| 07:18.15 | Guesty | I am trying to boost the quality of MOH, to be honest at the moment it sucks. Doesn't sound very 2017. I also tried 16 PCM 8000 wav with no better results, it only returns Unknown format. |
| 07:20.30 | Guesty | Even if I call internally it is bad quality |
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| 09:01.04 | _8bits | Hello, what does "Computed Login" or "Computed Logoff" means in ASTERNIC? |
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| 09:08.56 | _8bits | Hello, does anyone know what does "COMPUTED LOGIN/LOGOFF" means in ASTERNIC call center? |
| 09:16.47 | pawiecki | _8bits: show it |
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| 10:36.24 | jamesaxl | hi |
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| 14:21.15 | polysics | yo! |
| 14:21.25 | polysics | I wonder if anyone can help me troubleshoot a VERY weird issue |
| 14:23.55 | polysics | basically, I am trying to figure out if a pound key press is terminating my MRCPRecog instance because of some internals or if it's application code |
| 14:34.06 | thiagoc | Hi, anyone know how can I set a variable using de U option from Dial? Tried to use _ and __ but it doesn't work |
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| 15:14.27 | polysics | I am wondering what the best way is to see what "path" a DTMF digit is taking |
| 15:14.41 | polysics | as I do not really see it in the console but I do see it in the AMI output which is odd |
| 15:15.08 | Samot | Turn up DTMF logging. |
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| 15:30.22 | wonderworld | how would i run a shell command and get the output back into the dialplan? System() just seems to return SUCCESS or FAILURE. |
| 15:31.10 | Samot | Shell() |
| 15:32.17 | wonderworld | Samot: tnx |
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| 15:51.57 | polysics | Samot: this will sounds VERY stupid. I can't find DTMF logging settings :D |
| 15:56.59 | polysics | ok, add dtmf in logger.conf |
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| 17:21.32 | elcontrastador | Building Asterisk 15.0.0rc1 on Ubuntu 16.0.4.3LTS get the following error during make: ./libasteriskssl.so: undefined reference to `CRYPTO_num_locks' |
| 17:21.51 | elcontrastador | Googling has given me solid hit on this. |
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| 17:26.19 | elcontrastador | hasn't |
| 17:27.25 | [TK]D-Fender | #asterisk-dev |
| 17:27.41 | [TK]D-Fender | Unreleased branches are supported in the dedicated dev channel |
| 17:28.11 | elcontrastador | k...thx |
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| 22:22.13 | sibiria | i have a somewhat annoying problem with pjsip: it doesn't always automatically quality remote hosts upon startup |
| 22:22.55 | sibiria | sometimes it qualifies some of them, sometimes not and then they are unavailable until qualify_frequency happens |
| 22:25.07 | sibiria | (13.13-cert6) |
| 22:28.20 | sibiria | this happens whether i have five trunks set up, two, or even just a single one |
| 22:32.51 | sibiria | which trunks are qualified and which aren't at startup is random. no discernable pattern |
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