My Asterisk server is having several connectivity problems that can be traced back to the codec issue. These are the devices involved:
- Asterisk 1.4
- Sipura/Linksys ATAs with varied firmware
- Cisco AS-5300 voice gateway (used for VoIP-> POTS calls only)
I keep on getting a “No available media” (i.e.: there are no common codecs) messages from the Cisco even when I have tried every combination of codecs. Sometimes the call goes through but the voice is unidirectional (no NAT involved).
What I need is to understand better the nature of codec negotiation, so I can debug and track down the specific failures. Is there some documentation that describes the codec negotiation protocol? What debugging mode should I use?
What I use now is
sip show peers
sip show peer xyz
What else should I look into?