sbj sip rtp unknown codec 72 error on fxo output channel and sip output channel for calls in from sip soft phone
Hello: I am seeking to place calls from sip soft phone through asterisk to go out zap card to fxs device or go out fxo to regular phone line to the telco. On one machine all of this is working. On second machine, calls out to the zap channel hocked to standard phone are working. Calls out to zap channel, fxo, on phone line, are failing. From head phones to the sip soft phone at call start I hear a muffled sound then silence, mostly. Console screen gives Notice messages:
Jul 30 02:34:21 NOTICE: rtp.c:505 ast_rtp_read: Unknown RTP codec 72 received
Jul 30 02:34:31 NOTICE: rtp.c:505 ast_rtp_read: Unknown RTP codec 72 received
– Hungup 'Zap/3-1’
In source code rtp.c, the number 72 seems to come from PayloadType, and is a value higher than MaxRtpPldNumber
The telephony card is the tdm400p from Digium using option for 2 fxs, 2 fxo modules.
Gm alaw, ulaw, ilbc codecs are enabled in sip.conf. On one sip software phone all these are supported. On another s/w phone, support is reported for gsm, pcmu, ilbc.
On the pc used for comparison, the call in pc to phone direction behaves normally. What are your ideas for troubleshooting this?
As another test, sip pc to pc call, was performed. It also showed Unknown codec 72 notice in asterisk console window.
Make clean, make, make install command issued in /usr/src/asterisk has been done on the machine where the problems exist but there was no improvement, seeming to eliminate corrupted binaries prospect…
Sincerely - - octimotor - - 7/30/2005, 7/31/2005;