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

Sat Aug 4 00:56:46 MSD 2012

I'm talking about a specific channel variable that you say works when it's
set global but does not work when you only use the "set" app in the
dialplan. I recommend you take it out of vars.xml and then try using the
export application or put it in the dialstring using the
{chan_var=value}sofia/foo/bar at baz notation...

-MC

On Fri, Aug 3, 2012 at 1:49 PM, A E G <all.eforums at gmail.com> wrote:

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