[Freeswitch-users] Annex B

[Ken Fulmer]

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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, May 04, 2010 5:19 PM
To: freeswitch-users at 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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