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| 00:21.21 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); 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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| 09:57.43 | ziz212 | Hi |
| 09:58.36 | ziz212 | Please some one can help me to findout how can we disconnect one side of a call and connect other side to conference room |
| 10:00.28 | ziz212 | already created |
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| 13:26.45 | Samot | I always love the reaction of people when I tell them SwitchVox is wrong for their custom project. |
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| 17:17.18 | Dovid | Is there any way to reload a specific MOH class? |
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| 18:58.55 | Dovid | Is the login for jira.digium.com not connected to issues.asterisk.org? |
| 18:59.26 | file | jira.digium.com is the Digium corporate JIRA |
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| 19:09.37 | lorsungcu | Samot: but muh APIs |
| 19:12.47 | Samot | They wanted to do something with ColdFusion with it |
| 19:13.16 | lorsungcu | Samot: this is sounding eerily similar to someone i know |
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| 19:38.31 | Dovid | I just went fron 13.8 to asterisk 14.5 is this a bug? . |
| 19:38.39 | Dovid | https://pastebin.com/wzZcXNBn |
| 19:40.22 | file | https://issues.asterisk.org/jira/browse/ASTERISK-27026 |
| 19:42.06 | Dovid | file: Thanks. Anything I can do to help? my bt isnt showing the same trace |
| 19:42.21 | Dovid | mine is dying instantly on load |
| 19:42.37 | file | yes it's the same backtrace |
| 19:42.40 | file | and that is the same problem |
| 19:43.02 | file | and there is a workaround on the issue, ensure an empty ari.conf exists |
| 19:43.41 | Dovid | file: So please humor me. Why does my bt now show the same errors (about memory etc.) |
| 19:44.18 | file | I don't understand the question |
| 19:44.26 | Dovid | meaning why am I not showing the same output as him when I do a bt |
| 19:44.32 | Dovid | https://issues.asterisk.org/jira/secure/attachment/55522/backtrace.txt |
| 19:44.39 | file | because you did a different gdb command |
| 19:44.55 | Dovid | is the command: gdb /usr/sbin/asterisk CORE_FILE? |
| 19:44.58 | file | "thread apply all bt" |
| 19:44.59 | Dovid | i want to make sure I do it right |
| 19:45.00 | file | would show more |
| 19:45.46 | file | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationAfterACrash |
| 19:45.51 | Dovid | file: Thanks |
| 19:45.53 | file | has a full gdb command that gets it |
| 19:46.17 | Dovid | yup. i see that now. I need to hit up the wiki more often |
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| 19:54.10 | Dovid | file: I have a question on a possible bug. Looking at: https://issues.asterisk.org/jira/browse/ASTERISK-25974 if you have cachertclasses=yes then if say we have three calls using MOh Class my_test and we do a moh reload then any new call will start a new thread. when all calls using the old thread hangup it will die. shouldnt asterisk only purge moh if it's not in use? |
| 19:55.44 | file | I have no idea. |
| 19:56.10 | Dovid | ok. thankis |
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| 21:00.35 | XATRIX | Hi, can you advice ? Why i have no RTP traffic ? Seems like my 3G USB dongle does the call, but nothing i can hear while i'm calling. Only call reset tones after a while. |
| 21:00.39 | XATRIX | https://paste.fedoraproject.org/paste/QEhtYyPsaKKqXeDbcMeHxw |
| 21:00.58 | XATRIX | I have 5060 and RTP portrange forwarded on my firewall |
| 21:01.47 | XATRIX | also i have double NAT between my networks |
| 21:02.30 | Samot | And you have private IPs in your SDP on both sides of an external call. |
| 21:02.49 | Samot | So yeah, RTP is there but going to IPs that can't route over the Internet. |
| 21:03.29 | XATRIX | SIP_client 192.168.199.3(NAT)178.212.64.96 ----- 77.88.209.171(NAT)192.168.1.59 SIP_SERVER |
| 21:03.48 | Samot | c=IN IP4 192.168.1.59 |
| 21:03.59 | Samot | ^^ Not going to go over the Internet. |
| 21:04.17 | XATRIX | yea, i understand, but how to fix ? |
| 21:04.34 | Samot | Well make sure you SIP settings are correct. |
| 21:04.41 | XATRIX | I suppose, it should be c=IN IP4 77.88.209.171 |
| 21:04.51 | XATRIX | shoudn't it ? |
| 21:04.56 | Samot | And that your device has NAT enabled on it. |
| 21:05.11 | Samot | It should be a WAN IP if it's going out over the Internet. |
| 21:05.13 | XATRIX | give me a sec |
| 21:05.14 | Samot | On both sides. |
| 21:05.45 | Samot | Also you have FreePBX so you should be asking about there in there. |
| 21:05.55 | Samot | s/there/this/ |
| 21:08.01 | XATRIX | Samot, https://pasteboard.co/hoqGEhvMx.png |
| 21:08.16 | XATRIX | It's a SIP SERVER firewall port forward |
| 21:09.07 | XATRIX | But it's a general question, isn't it ? I'm not asking about freepbx settings. But if you say so... |
| 21:10.00 | file | that is a configured option of chan_sip, and FreePBX is what configures it and presents it in the GUI |
| 21:10.55 | XATRIX | alright, sorry. i'm going to ask there |
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