Wow, that worked great. Thanks so much...that's a lifesaver for us! Out of curiosity, where should I look in the documentation for this parameter? I didn't see it on the channel variable page. I may be looking in the wrong places and hate to keep asking you guys so many questions. Ken -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, May 04, 2010 5:19 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Annex B <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> /b On May 4, 2010, at 5:11 PM, Ken Fulmer wrote:
When we send calls to an Adtran gateway, our calls using g.729 work properly. I noticed in the SDP the following field:
a=fmtp:18 annexb=yes
When we send calls to a Cisco gateway, the same call fails.
Inbound g.729 calls coming from the Cisco gateway work properly and I noticed the following:
a=fmtp:18 annexb=no
Is there a way to set this parameter in the dial-plan or sip profiles so we can match what the Cisco gateway is looking for?
Thanks,
Ken Fulmer
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