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

[A E G]

On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc at freeswitch.org> wrote:

> 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
>
>
Solved it using the Global variable in vars.xml. Sticking
sdp_m_per_ptime=false as a channel variable (just before calling bridge)
didn't seem to have any affect on the behaviour. Didn't try it in the
specific profile to which that gateway belongs.


> 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
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
> Join Us At ClueCon - Aug 7-9, 2012
>
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