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

[A E G]

...and I'm not done yet.

So while this one below solved the problem, how do we manage this so I
don't always suppress all the m= lines in my SDP when sending calls to all
the external gateways, but do that only when sending calls to any system
that doesn't like it?

As I said before, I tried doing this just before the bridge, but didn't
work. Only seems to work as a PRE-PROCESS Global setting.

Thx

On Fri, Aug 3, 2012 at 1:44 AM, A E G <all.eforums at gmail.com> wrote:

> 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
>>
>>
>>
>>
>>
>>
>
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