SOLVED: Asterisk don't like multiple m= lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
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
...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
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 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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc@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@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@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@baz notation... -MC On Fri, Aug 3, 2012 at 1:49 PM, A E G <all.eforums@gmail.com> wrote:
On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc@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@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@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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On Fri, Aug 3, 2012 at 4:56 PM, Michael Collins <msc@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@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@gmail.com> wrote:
On Fri, Aug 3, 2012 at 4:26 PM, Michael Collins <msc@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@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@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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A E G
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Michael Collins