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

[Michael Collins]

Could be an issue of export vs. set...
-MC

On Fri, Aug 3, 2012 at 1:03 PM, A E G <all.eforums at gmail.com> wrote:

> 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
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> Join Us At ClueCon - Aug 7-9, 2012
>>
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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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