[Asterisk-Dev] SIP Format Bug??

[Gene Willingham]

SIP Invite Fragment

v=0
o=root 17505 17508 IN IP4 4.79.19.58
s=session
c=IN IP4 4.79.19.58
t=0 0
m=audio 19328 RTP/AVP 0 18 111 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=noa=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


I am trying to debug an issue with the handling of rfc2833 in Asterisk 1.2.
Whether rfc2833 works or not depends on which options are defined for the
peer.

The offending line is:

a=fmtp:18 annexb=noa=rtpmap:111 G726-32/8000

In this example rfc2833 works, because formatting error moves.  If I do not
specify g726, then it fails.  This is what the line looks like.

a=fmtp:18 annexb=noa=rtpmap:101 telephone-event/8000^M