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

[Michael Collins]

Did you solve it with setting a channel variable or a Sofia profile
setting? If a channel variable then you should be able to do a little
dialplan logic and set the value only if certain conditions are met, i.e.
if a certain gateway is going to be used.

-MC

On Fri, Aug 3, 2012 at 12:44 PM, A E G <all.eforums at gmail.com> wrote:

> ...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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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
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