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| 04:49.30 | *** topic/#asterisk is Take the March 2019 Asterisk User Survey! https://goo.gl/forms/xL1VUHRsf95saly13 -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.28.0 (2019/7/25) 16.5.0 (2019/7/25), Security Only: 15.7.3 (2019/7/11); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 05:44.58 | devdvd | @file samot figured it out. Feel like a bit of a moron for not seeing this now. Had an extra character at the top of pjsip.conf that was breaking the whole thing |
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| 07:24.16 | masked | thanks Samot and everyone for your help |
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| 07:54.57 | masked | Samot: or someone else.. could you take a look at this pastebin and tell me why it plays the unavailable message and then hangs up and doesn't forward to voicemail? |
| 07:55.00 | masked | https://pastebin.com/QHqt8aS2 |
| 07:55.16 | Reinhilde | I haven't looked but it's possible the voicemail box is not defined |
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| 08:00.50 | masked | Reinhilde: given im using odbc with a view in mysql maybe i need to add an extension in vmusers view? |
| 08:01.09 | masked | i kinda assumed opensips would do that when i created the user |
| 08:01.36 | masked | i'll have to install adminer or something and take a look |
| 08:03.55 | masked | voicemail show users for default |
| 08:04.00 | masked | shows it exists |
| 08:04.58 | masked | any other idieas Reinhilde ? |
| 08:05.19 | Reinhilde | does asterisk know the user in any capacity? |
| 08:05.38 | masked | no not to my knowledge |
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| 08:07.42 | Reinhilde | well that's required if you want it to collect voicemails for him |
| 08:08.11 | masked | news to me |
| 08:08.47 | Reinhilde | <PROTECTED> |
| 08:09.18 | Reinhilde | is the mailbox 61383720168 defined under voicemail.conf:[default] ? |
| 08:11.20 | masked | the odbcstorage and db thing is yes |
| 08:11.35 | Reinhilde | that does not answer the question. |
| 08:11.36 | masked | i think i need sipusers => odbc etc defined in res_odbc.conf |
| 08:11.36 | Reinhilde | is the mailbox 61383720168 defined under voicemail.conf:[default] ? |
| 08:11.44 | masked | no |
| 08:12.02 | Reinhilde | that would be your problem, unless voicemail => <database> is defined in whateverwherever |
| 08:13.30 | masked | voicemail => odbc,asterisk,vmusers is defined |
| 08:13.35 | masked | but sipusers and sippeers are not |
| 08:13.38 | masked | do they need to be? |
| 08:14.02 | masked | gtg getting off the train bbs |
| 08:14.16 | Reinhilde | they don't, masked |
| 08:14.24 | Reinhilde | but realtime is really unreliable |
| 08:16.51 | Reinhilde | ok, it should be going to voicemail. |
| 08:16.55 | Reinhilde | masked: try deleting the Hangup() |
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| 08:35.42 | masked | Reinhilde: ok ill try that |
| 08:38.23 | masked | Reinhilde: nope. same problem |
| 08:38.30 | Reinhilde | ugh |
| 08:38.35 | masked | it plays the number, says they are unavailable then hangs up |
| 08:38.41 | Reinhilde | that's very strange |
| 08:38.52 | Reinhilde | escalate it to the experts. |
| 08:40.06 | masked | who are the experts here? Samot ? |
| 08:40.28 | Reinhilde | So far. |
| 08:40.54 | masked | cool ill wait |
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| 09:21.47 | masked | okay fixed |
| 09:21.59 | masked | i ran make basic-pbx |
| 09:22.07 | masked | then copied my configs over the top |
| 09:22.32 | masked | i think it was looking for vm-intro.gsm but couldn't find it due to an incorrect env var path |
| 09:27.53 | masked | thanks again |
| 09:28.23 | masked | i like it when people still talk and are helpful on irc these days |
| 09:28.28 | masked | i think i'll stick around |
| 09:28.32 | masked | might install a bnc |
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| 09:55.51 | masked | alright great everything is working |
| 09:56.36 | masked | now i think if i prefix the ruri and add that to an extension followed by a . and then remove the prefix with $EXTEN:Number i should be able to parse all calls |
| 09:56.56 | masked | i can receive and listen to voicemails, everythign is sweet |
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| 12:53.32 | megaTherion | Hi |
| 12:54.23 | megaTherion | What's wrong with "exten => +49XXXXXXXXX,1,Goto(incoming,0XXXXXXXXX,1)"? - I've tried to route incoming calls to this number - but it doesnt work this way. :( |
| 12:57.03 | megaTherion | that 'X' is the complete number... which is correct ,the one without +49 actually works just fine |
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| 12:58.37 | Samot | You'll need to show it happening. |
| 12:59.34 | megaTherion | I see, how? |
| 12:59.45 | megaTherion | with asterisk -r on the CLI? |
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| 13:06.34 | Samot | asterisk -rvvvvvvvvvv |
| 13:06.39 | Samot | then make the call |
| 13:07.23 | megaTherion | ok ;) |
| 13:07.25 | megaTherion | I'll try |
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| 13:22.32 | mn3monic | hello, any programmer experienced with pjsip willing to answer a stackoverflow question ? :-) https://stackoverflow.com/questions/57738352/unable-to-do-two-calls-and-conference-with-python-pjsip |
| 13:22.33 | mn3monic | thanks |
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| 19:42.41 | masked | cool. i setup a prefix in opensips and dropped it in asterisk now it answers all users/extensions :) |
| 19:42.51 | masked | very happy :) |
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