Could be an issue of export vs. set... -MC On Fri, Aug 3, 2012 at 1:03 PM, A E G <all.eforums@gmail.com> wrote:
On Fri, Aug 3, 2012 at 3:51 PM, Michael Collins <msc@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@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@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@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@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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