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

Sat Aug 4 01:08:27 MSD 2012

On Fri, Aug 3, 2012 at 4:56 PM, Michael Collins <msc at freeswitch.org> wrote:

> 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
>
>
Ok, so that's what I was going to ask; is setting the channel variable in
the dialstring different than setting it within the dialplan just before
bridge is called. I guess Leg B doesn't really start until the Bridge is
called ...duh! Setting the {chan_var} in the dialstring worked. Thanks!

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