[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Kevin P. Fleming]

Aaron Clauson wrote:

> m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:100 NSE/8000

I don't know what this (NSE) is, but Asterisk certainly doesn't support it.

The only way we can debug this is by getting a complete 'sip debug' and 
'set verbose' console trace; read the bug posting guidelines at 
bugs.digium.com and open a bug there with the required information.