[Freeswitch-users] Annex B

[Brian West]

<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
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org