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| 05:03.53 | rpifan | i cant seem to get custom cdrs to work |
| 05:03.57 | rpifan | ive followed all the guides |
| 05:04.04 | rpifan | and yet the cdr-custom directory is still empty |
| 05:23.43 | rpifan | it logs fine to the regular master.csv |
| 05:23.50 | rpifan | but i cant get it to log to the custom csv |
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| 06:17.39 | [TK]D-Fender | ok... |
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| 09:19.50 | danielyk | Hi, I have problems with migrating from chan_sip to chan_pjsip. My provider is Deutsche Telekom. Here is my old REGISTER with chan_sip, there were never problems. http://pastebin.ca/3729118 Now I try pjsip and get sometimes 403 Errors as answer, if I register multiple times in follow (e.g. restarting asterisk) or changing the public ip (restart router). Here is the pjsip REGISTER: http://pastebin.ca/3729116 Does anyone see in the |
| 09:19.50 | danielyk | <PROTECTED> |
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| 13:26.33 | Glanzmann | Hello; with https://www.mizu-voip.com/portals/0/Images/asterisk_webrtc.png and https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 can I use sip over one single port or is RTP still going outside of the connectio TLS 443 connection? |
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| 14:10.21 | Glanzmann | To answer my own question. Websocket is used as sip proxy but rtp still uses udp in a seperate srtp connection. |
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| 18:19.13 | Glanzmann | Hello; I recently switched to pjsip and would like to use the call recording button on my snom. But I don't see any reaction when I press the button after I switched to pjsip. Any ideas? |
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