[Asterisk-Users] codec trouble?

[Evert Meulie]

Hi everyone!

Situation: when I call from cell phone to a asterisk-connected phone, 
all works fine. When I call from the asterisk-connected phone (a Cisco 
7960 SIP) to the cell, the connection gets made, but there is no audio 
going in either way...
Asterisk reports the following:
Sep 16 08:27:41 WARNING[245775]: chan_sip.c:2679 process_sdp: 
Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123')
Sep 16 08:27:47 WARNING[245775]: chan_sip.c:2679 process_sdp: 
Insufficient information for SDP (m = '', c = 'IN IP4 123.123.123.123')
Sep 16 08:27:47 WARNING[360465]: rtp.c:1382 ast_rtp_bridge: codec0 = 268 
is not codec1 = 0, cannot native bridge.
  == Spawn extension (sip, 88888888, 1) exited non-zero on 'SIP/105-1559'

(123.123.123.123 is the IP of our VoIP-provider, 88888888 is my cell 
phone, and 105 is the asterisk-connected phone).



Regards,
    Evert