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| 07:17.26 | oej | Morning folks |
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| 08:16.41 | Demon_VoIP | If I create two ACL in pjsip.conf: one that allows all IP, and the second, which prohibits all but one of the IP. |
| 08:16.41 | Demon_VoIP | Do I understand correctly that no registration will not be rejected? |
| 08:17.39 | oej | That was hard to parse : "no registrations will not be rejected" |
| 08:18.00 | oej | allow all, then allow only one only that IP will succeed |
| 08:18.03 | Demon_VoIP | Do I understand correctly that no registration will be rejected? :) Google helps bad :) |
| 08:18.30 | Demon_VoIP | thanks |
| 08:19.39 | Demon_VoIP | In this way I can use ACL from pjsip.conf, not from acl.conf. https://issues.asterisk.org/jira/browse/ASTERISK-24890 |
| 08:20.10 | Demon_VoIP | But... it is workaround. bad decision |
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| 08:59.18 | skrusty | morning |
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| 09:11.57 | freemanls | Hello guys, I have an audio loss in the middle of the call, could it be from NTP not configured on Cisco SIP phones? |
| 09:22.19 | Demon_VoIP | time and timer - different words :) I think "no". And you can try configure NTP and make call :) |
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| 09:27.51 | freemanls | :) |
| 09:28.02 | freemanls | it usually happens while going through an IVR menu |
| 10:02.21 | Demon_VoIP | codec slin? |
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| 10:03.16 | Demon_VoIP | How to force renegotiation of codecs do after IVR? |
| 10:20.48 | freemanls | Demon_VoIP: it is just an IVR call |
| 10:29.08 | Demon_VoIP | ok. It's another question :) An incoming call reaches the IVR or prompt (codecs alaw-> slin), after selecting an outgoing call occurs. The destination also supports alaw. Why in this case remains recoding via slin? Or in pjsip it fexed? |
| 10:30.32 | freemanls | oh sorry |
| 10:30.32 | freemanls | :D |
| 10:37.34 | Elio19 | Hey, i have a question too. How to do webRTC? I tried just about everything ( except pjsip ) I even took an 11.5 and patched it for opus, etc i was editing . c files by hand etc |
| 10:38.45 | Elio19 | TestPhone-A@crypotbiz.directory pass: TestPassDev- |
| 10:41.44 | Elio19 | Dial 555 for a recording, as soon as asterisk start Playback, the webrtc client quits with message: Mad Media Descriptor, which i think relates to: a=rtpmap:109 opus/48000/2 |
| 10:44.51 | Elio19 | I'm presently running 13.2 , and after successfully compile, install, and configure, i just want it to work |
| 10:47.07 | oej | "Mad Media Descriptor" sounds fun. What's next? "hysterical media descriptor" :-) |
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| 11:28.45 | skrusty | morning |
| 11:29.26 | freemanls | morning |
| 11:54.04 | Demon_VoIP | afternoon :) |
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