[Freeswitch-users] SOLVED: Asterisk don't like multiple m= lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]

A E G all.eforums at gmail.com
Fri Aug 3 09:44:44 MSD 2012


Well turns out learning to use Google better always helps.

Found a nugget of wisdom that <action application="set"
data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml

Seems to have fixed it.



On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums at gmail.com> wrote:

> Gents,
>
> running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a
>
> have been fighting with this for an hour or more...have done a bit of
> research on the list and Google itself, scoured the Wiki etc. but can't
> seem to figure out where to set the codecs to be "well recd" by the remote
> SIP peer. This used to work fine until I upgraded 2 days ago and it stopped
> working without my doing / changing anything.
>
> The remote SIP peer is {*} which doesn't like multiple m= lines with
> differing ptime values.
>
> Debug on near-end says: "Originate Resulted in Error Cause: 88
> [INCOMPATIBLE_DESTINATION]"
>
> Full SDP here:
>
> v=0
> o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80
> s=FreeSWITCH
> c=IN IP4 192.168.1.80
> t=0 0
> m=audio 28884 RTP/AVP 98 8 3 101 13
> a=rtpmap:98 L16/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=audio 28884 RTP/AVP 0 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:40
> a=sendrecv
>
> Far end says:  Rejecting non-primary audio stream: audio 28884 RTP/AVP 0
> 101 13
>
> I have tried to play around with the codec globals in vars.xml, to no
> avail.
>
> have also added stuff directly in the dialplan like so:
>
> <action application="set" data="hangup_after_bridge=true"/>
> <action application="set" data="absolute_codec_string=PCMU at 40i"/>
> <action application="set" data="sdp_m_per_ptime=false"/>
> <action application="bridge" data="sofia/gateway/mycore1/1111"/>
>
>
> but no dice.
>
> Tried to remove the "absolute_codec_string", still no dice.
>
> I read about the "sdp_m_per_ptime"  that lets me "cheat" but that ain't
> doing anything either.
>
> What gives?
>
> Thx in advance
>
>
>
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120803/c5f21827/attachment.html 


Join us at ClueCon 2011 Aug 9-11, 2011
More information about the FreeSWITCH-users mailing list