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| 09:19.44 | *** 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: 13.27.0 (2019/5/30) 16.4.0 (2019/5/30), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 10:55.41 | yang | Anyone using Localphone provider? I have a problem receiving incoming calls... |
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| 11:08.40 | yang | Also the outgoing calls seem to pick a random outgoing number |
| 11:12.46 | yang | http://paste.debian.net/plainh/20b5a59e |
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| 11:37.41 | yang | I solved setting up a caller ID number |
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| 14:47.52 | yang | What to do with ongoing SIP spam in Asterisk ? chan_sip.c:26273 handle_request_invite: Call from ... |
| 14:53.38 | yang | I placed "allowguest=no" in context [general] |
| 15:08.39 | sibiria | that would be the usual way, yes |
| 15:13.35 | flok | I tried the opposite; allowing guests and sending them to a conference-group, but that didn't work: https://pastebin.com/nrCPM6A4 - an ideas? |
| 15:25.04 | sibiria | flok: yes, it's because they don't try to call "s". they're calling an actual number, so you need to pattern-match for anything |
| 15:25.26 | sibiria | the typical approach is: |
| 15:25.46 | sibiria | exten => _x. |
| 15:26.01 | sibiria | and ditto for _+x to handle "+e.164" |
| 15:26.32 | sibiria | _+x. i mean |
| 15:28.28 | flok | sibiria: so _x. is a catch-all so to say? |
| 15:28.38 | sibiria | _x means any single digit |
| 15:28.46 | sibiria | _x+ means any consecutive number of digits |
| 15:28.52 | sibiria | _ <- enter pattern matching mode |
| 15:28.59 | sibiria | x = any digit 0-8 |
| 15:29.00 | sibiria | 0-9* |
| 15:29.56 | sibiria | sorry . (period) means "one or more" |
| 15:30.01 | sibiria | i'm stuck in regular regexp mode |
| 15:30.25 | flok | sibiria: thanks! |
| 15:30.47 | sibiria | there should be an article on asterisk's pattern matching on the wiki |
| 15:31.15 | sibiria | it's sort of a simplistic/minimal approach to regular expressions |
| 15:32.19 | sibiria | it's a funny trick that, trapping people and bridging them |
| 15:32.31 | flok | found it: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 15:32.33 | sibiria | don't forget to limit the number of members in that bridge, just so no one bogs down your server :) |
| 15:32.41 | sibiria | max_members is the option unless i misremember |
| 15:33.08 | flok | sibiria: yeah it is for the hackerspace. we have that bridge to have fun. we have phones in every room of the space. |
| 15:33.18 | flok | all with asterisk of course\ |
| 15:33.23 | flok | works like a charm |
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| 15:36.45 | flok | well, not the capture-all unforutnately. I tried "exten => _x+,1,Answer()" with _x+, _+x. _x. |
| 15:37.23 | sibiria | _x. <- will match any series of digits-only |
| 15:37.33 | sibiria | _+x. ditto when the number starts with + |
| 15:38.03 | sibiria | . is the "one or more" modifier in asterisk's pattern matching |
| 15:38.13 | sibiria | not + as i mistakenly said above - that's for "normal" regular expressions |
| 15:38.55 | sibiria | so, thusly: exten => _x.,1,Answer |
| 15:39.22 | sibiria | and then the same set of extensions but for _+x. just in case someone requests an extension starting with + |
| 15:41.08 | Reinhilde | explodify |
| 15:45.50 | flok | nope, call ended with no such context. I'm puzzled. could it be a nat issue? hmmmm. |
| 15:46.05 | sibiria | check what your logs say |
| 15:46.10 | sibiria | and, you did reload the dial plan after the change, right? |
| 15:46.21 | sibiria | make sure you point to the correct default context, too |
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| 15:46.43 | sibiria | in [general] that would be context=something |
| 15:46.57 | sibiria | it sounds like are using chan_sip, at least |
| 15:48.43 | flok | sibiria: I did /etc/init.d/asterisk reload, so yes. but it looks like a nat-issue from my client-end |
| 15:49.01 | flok | Channel SIP/192.168.65.220-00000019 - from ~wEh2queex |
| 15:49.14 | sibiria | "no such context" sounds to me like what you point at with 'context' doesn't exist in extensions.conf |
| 15:51.06 | flok | it must be zoiper doing something wrong because linphone connects fine just without sound due to that not-natting error (altough I configured a stun server in it) |
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| 15:54.14 | flok | 1234@space.nurdspace.nl I tried |
| 15:54.53 | sibiria | take a look at the logs to see what zoiper is actually requesting |
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| 15:57.01 | flok | it's not even reaching it. let me try it from my mobile phone |
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| 16:07.00 | *** 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: 13.27.0 (2019/5/30) 16.4.0 (2019/5/30), Security Only: 15.7.2 (2019/2/28); DAHDI: 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 17:32.38 | flok | what can be the cause if a asterisk sees a caller and starts streaming audio but the other side (a caller behind an asterisk server) doesn't receive the call-setupack? (and thus hears the ring-tone) |
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| 17:47.14 | [TK]D-Fender | Not getting sent to the right place |
| 17:47.21 | [TK]D-Fender | Or getting firewalled |
| 17:47.28 | [TK]D-Fender | basically exacly what you would imagine |
| 17:47.39 | [TK]D-Fender | Not sent to the right place or it got blocked. |
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| 20:29.55 | edman80 | Hello all. Voicemail.conf, is there any way to pass the audio to a pager message similar to how it is done with email messages? |
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