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directrtp=on - no audio · Issue #405 · chan-sccp/chan …

    https://github.com/chan-sccp/chan-sccp/issues/405
    Skinny Client Control Protocol (SCCP). Release: 4.3.2 (HEAD - 558d2b6M) As far as I can tell DirectRTP between SCCP endpoints on asterisk (7912/7940/7960) working correctly. But there is No DirectRTP in call between SCCP and PJSIP endpoint. RTP flow goes through the PBX for each call leg. Is it possible at all, in case of different signaling ...

No audio on Asterisk SIP call - Stack Overflow

    https://stackoverflow.com/questions/5554837/no-audio-on-asterisk-sip-call
    No audio on Asterisk SIP call. Ask Question Asked 10 years, 10 months ago. Active 1 year, 10 months ago. Viewed 34k times 9 I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not ...

No audio on chan_skinny 7941G - Asterisk Support ...

    https://community.asterisk.org/t/no-audio-on-chan-skinny-7941g/37040
    Hi all, as many others, first post 😃 I’ve got some problem with my Cisco 7941G and Asterisk (1.8.7.0). Before listing my problem, I want to explicit that I have a Cisco 7941G (firmware: SCCP41.8-4-2S), a SIP phone (Polycom IP320 SIP or Zoiper Free, but this is not influent) and Asterisk on a Ubuntu Server. Below there is my skinny.conf: [general] disallow=all …

SIP -> Local -> SCCP one way audio because of wrong …

    https://github.com/chan-sccp/chan-sccp/issues/420
    Install revision c461215. Make a call with local channel from a sip phone with (PJSIP) allow: (g722|alaw|ulaw|opus|gsm|g726|g729) and with g722 as first priority in the invite. Asterisk is then sending G722 to the 7965 and no sound. Expected behavior: Before c461215 there was sound in both directions.

How to troubleshoot one-way / no audio issues - Cisco ...

    https://community.cisco.com/t5/collaboration-voice-and-video/how-to-troubleshoot-one-way-no-audio-issues/ta-p/3164442
    How to check really quick if the phones are sending / receiving RTP (audio). * Open the web page for 2 test phones, then click the 'stream 1' link located at the left handed side of the page, and check if the IP address and port match the information on both sides, keep pressing the 'stream 1' link and you will notice that the Tx and Rx stats ...

RasPBX - Asterisk for Raspberry Pi / Discussion / General ...

    https://sourceforge.net/p/raspbx/discussion/general/thread/50bc3f3f/
    Need help on getting SCCP working. Bought a Cisco 7921 and thought it was SIP (huh). Running Raspbx / Asterisk 11.10.0 & FreePBX 2.11.0.23. I have been able to get the telephone to register and i can place calls but as said - no audio in any direction. Using default skinny.conf with some small changes.

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