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| 06:27.12 | nunne | Getting "tcptls.c:222 handle_tcptls_connection: Certificate did not verify: unable to get local issuer certificate" when a Lync server tries to connect to my asterisk 11 server. Anyone know what might be wrong? It's a cert from godaddy (i know...). But it loads OKAY "SSL Certificate ok" when doing SIP reload. What am I missing here? I even asked the people |
| 06:27.12 | nunne | that are managing the Lync server to add the godaddy intermediate certificate just to be sure. |
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| 11:46.06 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 13:16.36 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 14:34.35 | Stefan27 | in manager.conf how do I use eventfilter to only allow certain events say PeerStatus only |
| 14:34.41 | boom^time | Can anyone recommend a good way to determine if an ANI is a wireless number or not? |
| 14:39.48 | [TK]D-Fender | boom^time: You'll need a list from every region you're looking at for that |
| 14:39.54 | Stefan27 | nvm I understood the documentation correct after all, just didn't restart properly |
| 14:40.54 | boom^time | [TK]D-Fender, I'm reading that the LRN database can provide me with that. |
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| 15:02.27 | StaRetji | howdy |
| 15:03.25 | StaRetji | I have diaplan which should dial extension and start Playback when extension answers, but, it actually start playback after Dial |
| 15:04.08 | StaRetji | exten => 100,1,Dial(SIP/101@TRT,20) |
| 15:04.44 | StaRetji | exten => Playback(howdy) |
| 15:04.52 | [TK]D-Fender | that's wrong on several levels |
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| 15:05.06 | [TK]D-Fender | dialplan does NOT continue processing after they answer |
| 15:05.10 | [TK]D-Fender | You should know better |
| 15:05.34 | StaRetji | sorry, typo |
| 15:05.37 | [TK]D-Fender | For things that can happen DURING that Dial... go read its instructions. |
| 15:05.43 | [TK]D-Fender | "core show application dial" |
| 15:05.55 | StaRetji | exten => 100,n,Playback(howdy) |
| 15:06.02 | StaRetji | ok |
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| 15:10.05 | StaRetji | so, there should be GotoIf ${DIALSTATUS}=ANSWER or something? |
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| 15:10.48 | [TK]D-Fender | [11:05][TK]D-FenderFor things that can happen DURING that Dial... go read its instructions. |
| 15:14.44 | StaRetji | Unless there is a timeout specified, the Dial application will wait |
| 15:14.44 | StaRetji | indefinitely until one of the called channels answers |
| 15:15.04 | Qwell | StaRetji: answer, among several other options |
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| 15:34.25 | DynamicFail | How difficult would it be to create a dialplan that would call two different numbers (each of which may be a conference on a different non-asterisk pbx that I wouldn't control) then merge both calls into a conference? |
| 15:34.32 | DynamicFail | well I guess is it even possible |
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| 15:39.08 | [TK]D-Fender | DynamicFail: easy. |
| 15:39.15 | [TK]D-Fender | "core show application dial" |
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| 15:56.55 | smkelly | file: hi |
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| 16:01.14 | smkelly | file: sup |
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| 19:35.16 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.12.0 (2014/08/19), 1.8.30.0 (2014/08/19); Standard: Asterisk 12.5.0 (2014/08/19); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 20:53.13 | masus | hi, i need set some variables on answer RetryDial cmd, i use G(context^exten^pri) but the call hangup after execution. is there any solution for something like this? |
| 20:56.46 | [TK]D-Fender | masus: Show us |
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| 21:07.38 | [TK]D-Fender | \o/ |
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| 21:12.25 | masus | [TK]D-Fender: http://pastebin.com/mEdbvmJ2 |
| 21:12.49 | [TK]D-Fender | massthe FULL CALL |
| 21:13.20 | [TK]D-Fender | masus: And the dialplan to back it up |
| 21:14.45 | masus | [TK]D-Fender: http://pastebin.com/sggG3JH2 |
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| 21:18.02 | masus | [TK]D-Fender: i need to set some variables and play an anouncement to the caller and callee on answer the call. |
| 21:19.37 | masus | [TK]D-Fender: playing the anouncement and set the variable is working fine but the call hangup after the anouncement. |
| 21:19.52 | [TK]D-Fender | masus: "but the call hangup after execution" <- of course it hangs up. If you read Dial's instructions you would see that it does not say "this call will get bridged" at any point |
| 21:20.19 | [TK]D-Fender | masus: It does a GOTO on both channels and that is ALL. No more communication between the two. |
| 21:21.25 | masus | [TK]D-Fender: can u point me to the right direction? |
| 21:22.26 | [TK]D-Fender | masus: Read Dial's other options. Or consider using BRIDGE to hook them back up again, or a conference. |
| 21:23.27 | masus | [TK]D-Fender: ok thank you. i'll give it a try. |
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