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

Sat Aug 4 00:49:13 MSD 2012

On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc at freeswitch.org> wrote:

> Could be an issue of export vs. set...
> -MC
>
>
The Wiki says that export allows to carry over the value of a variable from
Leg A to Leg B. I am not sure this will solve my issue as I don't want to
export the codec offerings, more specifically varying ptime from Leg A to
Leg B. For this particular gateway, if I can, I only want to send
PCMU at 40i.Sorry if this is a stupid question, am an FS novice :)

>
> 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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>>>> 
>>>> 
>>>>
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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
>>>
>>> 
>>> 
>>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
>> 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
>
>
>
> _________________________________________________________________________
> 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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