Changing codec during calls
Hi all, How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. Cheers and thanks, Emrah
What happened with the uuid_media_reneg command? -MC On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote:
Hi all,
How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722.
Cheers and thanks, Emrah _________________________________________________________________________ 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
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
Hey, In short, nothing seems to happen. I tried several phones. I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it. This is what happens: freeswitch@internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, 1 total. freeswitch@internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU +OK Success 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 17888 RTP/AVP 9 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [calling][0] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000@sip.domain.net Update Callee ID to "10000" <10000> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [completing][200] 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [ready][200] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 Thanks for any help, Emrah On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> wrote:
What happened with the uuid_media_reneg command? -MC
On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: Hi all,
How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722.
Cheers and thanks, Emrah _________________________________________________________________________ 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
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
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
Try: uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU (ie add = before PCMU) It appears to affect something in the code... -Steve On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote:
Hey,
In short, nothing seems to happen. I tried several phones. I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it.
This is what happens: freeswitch@internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,,
1 total.
freeswitch@internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU +OK Success
2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 17888 RTP/AVP 9 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [calling][0] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000@sip.domain.net Update Callee ID to "10000" <10000> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [completing][200] 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [ready][200] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101
Thanks for any help, Emrah On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> wrote:
What happened with the uuid_media_reneg command? -MC
On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: Hi all,
How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722.
Cheers and thanks, Emrah _________________________________________________________________________ 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
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
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
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Excellent, that worked. However I am having a one way audio issue, where the peer that has its codecs changed can hear but cannot be heard. Even if both bridged peers are changed to a common codec it doesn't work. Any idea? Thanks a bunch for your help. On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote:
Try: uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU
(ie add = before PCMU)
It appears to affect something in the code...
-Steve
On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote:
Hey,
In short, nothing seems to happen. I tried several phones. I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it.
This is what happens: freeswitch@internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,,
1 total.
freeswitch@internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU +OK Success
2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 17888 RTP/AVP 9 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [calling][0] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000@sip.domain.net Update Callee ID to "10000" <10000> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [completing][200] 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [ready][200] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101
Thanks for any help, Emrah On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> wrote:
What happened with the uuid_media_reneg command? -MC
On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: Hi all,
How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722.
Cheers and thanks, Emrah _________________________________________________________________________ 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
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
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
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
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
Great. The code seems to show that the default is to append it to the current codec list, but = replaces the current codec list instead. I'll add a comment to that effect to the Wiki. As far the bridged peers goes, if you do a packet trace of the RTP do you see media with the correct codec being sent to the client? That might shed light on whether it's no media sent, wrong codec, or the some problem/bug on the peer. Steve On 19 February 2013 16:37, Emrah <lists@kavun.ch> wrote:
Excellent, that worked.
However I am having a one way audio issue, where the peer that has its codecs changed can hear but cannot be heard. Even if both bridged peers are changed to a common codec it doesn't work.
Any idea?
Thanks a bunch for your help. On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote:
Try: uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU
(ie add = before PCMU)
It appears to affect something in the code...
-Steve
On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote:
Hey,
In short, nothing seems to happen. I tried several phones. I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it.
This is what happens: freeswitch@internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k,XML,ek-conference,G722,16000,64000,G722,16000,64000,,cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,,
1 total.
freeswitch@internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU +OK Success
2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 17888 RTP/AVP 9 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [calling][0] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/10000@sip.domain.net Update Callee ID to "10000" <10000> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [completing][200] 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [ready][200] 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP v=0 o=- 3570241475 3570241476 IN IP4 10.0.0.131 s=pjmedia c=IN IP4 10.0.0.131 t=0 0 a=X-nat:0 m=audio 4010 RTP/AVP 9 101 a=rtcp:4011 IN IP4 10.0.0.131 a=rtpmap:9 G722/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101
Thanks for any help, Emrah On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> wrote:
What happened with the uuid_media_reneg command? -MC
On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: Hi all,
How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722.
Cheers and thanks, Emrah _________________________________________________________________________ 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
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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@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
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
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
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
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
pastebin the log. I'll bet Steven can diagnose it! ;) -MC On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists@kavun.ch> wrote: > Excellent, that worked. > > However I am having a one way audio issue, where the peer that has its > codecs changed can hear but cannot be heard. > Even if both bridged peers are changed to a common codec it doesn't work. > > Any idea? > > Thanks a bunch for your help. > On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote: > > > Try: > > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU > > > > (ie add = before PCMU) > > > > It appears to affect something in the code... > > > > -Steve > > > > > > On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote: > >> Hey, > >> > >> In short, nothing seems to happen. I tried several phones. > >> I also tried enabling codec renegotiation on hold/unhold thinking that > the change in SDP with uuid_media_reneg would do it. > >> > >> This is what happens: > >> freeswitch@internal> show channels > >> > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num > >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 > 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah > - Macbook > Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k > ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, > cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, > >> > >> 1 total. > >> > >> freeswitch@internal> uuid_media_reneg > 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU > >> +OK Success > >> > >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: > >> v=0 > >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 > >> s=FreeSWITCH > >> c=IN IP4 1.2.3.4 > >> t=0 0 > >> m=audio 17888 RTP/AVP 9 0 101 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16 > >> a=silenceSupp:off - - - - > >> a=ptime:20 > >> a=sendrecv > >> > >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000@sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/ > 10000@sip.domain.net entering state [calling][0] > >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000@sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000@sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ > 10000@sip.domain.net Update Callee ID to "10000" <10000> > >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/ > 10000@sip.domain.net entering state [completing][200] > >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: > >> v=0 > >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 > >> s=pjmedia > >> c=IN IP4 10.0.0.131 > >> t=0 0 > >> a=X-nat:0 > >> m=audio 4010 RTP/AVP 9 101 > >> a=rtcp:4011 IN IP4 10.0.0.131 > >> a=rtpmap:9 G722/8000 > >> a=sendrecv > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-15 > >> > >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000@sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send > signal sofia/internal/10000@sip.domain.net [BREAK] > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/ > 10000@sip.domain.net entering state [ready][200] > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP > >> v=0 > >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 > >> s=pjmedia > >> c=IN IP4 10.0.0.131 > >> t=0 0 > >> a=X-nat:0 > >> m=audio 4010 RTP/AVP 9 101 > >> a=rtcp:4011 IN IP4 10.0.0.131 > >> a=rtpmap:9 G722/8000 > >> a=sendrecv > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-15 > >> > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec > Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 > >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf > send/recv payload to 101 > >> > >> > >> Thanks for any help, > >> Emrah > >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> > wrote: > >> > >>> What happened with the uuid_media_reneg command? > >>> -MC > >>> > >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: > >>> Hi all, > >>> > >>> How do we go about renegotiating the codecs of a call that is already > established? > >>> I tried uuid_media_reneg to no avail. > >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. > >>> > >>> Cheers and thanks, > >>> Emrah > >>> > _________________________________________________________________________ > >>> 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 > >>> > >>> 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 > >>> > >>> 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 > >> > >> 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 > > > > 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 > > 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
Wow! This great feature is really work. I have the same issue - one way audio. And another: "show channels" always return first negotiated codec. When switching codec back, audio working fine. Tested with 3 different softphones. If someone interested, please confirm. Thanks. 2013/2/19 Michael Collins <msc@freeswitch.org> > pastebin the log. I'll bet Steven can diagnose it! ;) > -MC > > > On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists@kavun.ch> wrote: > >> Excellent, that worked. >> >> However I am having a one way audio issue, where the peer that has its >> codecs changed can hear but cannot be heard. >> Even if both bridged peers are changed to a common codec it doesn't work. >> >> Any idea? >> >> Thanks a bunch for your help. >> On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote: >> >> > Try: >> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >> > >> > (ie add = before PCMU) >> > >> > It appears to affect something in the code... >> > >> > -Steve >> > >> > >> > On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote: >> >> Hey, >> >> >> >> In short, nothing seems to happen. I tried several phones. >> >> I also tried enabling codec renegotiation on hold/unhold thinking that >> the change in SDP with uuid_media_reneg would do it. >> >> >> >> This is what happens: >> >> freeswitch@internal> show channels >> >> >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >> 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah >> - Macbook >> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k >> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >> cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, >> >> >> >> 1 total. >> >> >> >> freeswitch@internal> uuid_media_reneg >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >> >> +OK Success >> >> >> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >> >> v=0 >> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >> >> s=FreeSWITCH >> >> c=IN IP4 1.2.3.4 >> >> t=0 0 >> >> m=audio 17888 RTP/AVP 9 0 101 >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-16 >> >> a=silenceSupp:off - - - - >> >> a=ptime:20 >> >> a=sendrecv >> >> >> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000@sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/ >> 10000@sip.domain.net entering state [calling][0] >> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000@sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000@sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >> 10000@sip.domain.net Update Callee ID to "10000" <10000> >> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/ >> 10000@sip.domain.net entering state [completing][200] >> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >> >> v=0 >> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> >> s=pjmedia >> >> c=IN IP4 10.0.0.131 >> >> t=0 0 >> >> a=X-nat:0 >> >> m=audio 4010 RTP/AVP 9 101 >> >> a=rtcp:4011 IN IP4 10.0.0.131 >> >> a=rtpmap:9 G722/8000 >> >> a=sendrecv >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-15 >> >> >> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000@sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >> signal sofia/internal/10000@sip.domain.net [BREAK] >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/ >> 10000@sip.domain.net entering state [ready][200] >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >> >> v=0 >> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> >> s=pjmedia >> >> c=IN IP4 10.0.0.131 >> >> t=0 0 >> >> a=X-nat:0 >> >> m=audio 4010 RTP/AVP 9 101 >> >> a=rtcp:4011 IN IP4 10.0.0.131 >> >> a=rtpmap:9 G722/8000 >> >> a=sendrecv >> >> a=rtpmap:101 telephone-event/8000 >> >> a=fmtp:101 0-15 >> >> >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >> send/recv payload to 101 >> >> >> >> >> >> Thanks for any help, >> >> Emrah >> >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> >> wrote: >> >> >> >>> What happened with the uuid_media_reneg command? >> >>> -MC >> >>> >> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: >> >>> Hi all, >> >>> >> >>> How do we go about renegotiating the codecs of a call that is already >> established? >> >>> I tried uuid_media_reneg to no avail. >> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >> G722. >> >>> >> >>> Cheers and thanks, >> >>> Emrah >> >>> >> _________________________________________________________________________ >> >>> 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 >> >>> >> >>> 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 >> >>> >> >>> 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 >> >> >> >> 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 >> > >> > 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 >> >> 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 > > 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 > >
Definitely get the console log and SIP trace of those problem calls and put them in pastein.freeswitch.org. -MC On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com>wrote: > Wow! This great feature is really work. I have the same issue - one way > audio. And another: "show channels" always return first negotiated codec. > When switching codec back, audio working fine. Tested with 3 different > softphones. > If someone interested, please confirm. > Thanks. > > > 2013/2/19 Michael Collins <msc@freeswitch.org> > >> pastebin the log. I'll bet Steven can diagnose it! ;) >> -MC >> >> >> On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists@kavun.ch> wrote: >> >>> Excellent, that worked. >>> >>> However I am having a one way audio issue, where the peer that has its >>> codecs changed can hear but cannot be heard. >>> Even if both bridged peers are changed to a common codec it doesn't work. >>> >>> Any idea? >>> >>> Thanks a bunch for your help. >>> On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote: >>> >>> > Try: >>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>> > >>> > (ie add = before PCMU) >>> > >>> > It appears to affect something in the code... >>> > >>> > -Steve >>> > >>> > >>> > On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote: >>> >> Hey, >>> >> >>> >> In short, nothing seems to happen. I tried several phones. >>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>> that the change in SDP with uuid_media_reneg would do it. >>> >> >>> >> This is what happens: >>> >> freeswitch@internal> show channels >>> >> >>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>> 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah >>> - Macbook >>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k >>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>> cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, >>> >> >>> >> 1 total. >>> >> >>> >> freeswitch@internal> uuid_media_reneg >>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>> >> +OK Success >>> >> >>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>> >> v=0 >>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>> >> s=FreeSWITCH >>> >> c=IN IP4 1.2.3.4 >>> >> t=0 0 >>> >> m=audio 17888 RTP/AVP 9 0 101 >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-16 >>> >> a=silenceSupp:off - - - - >>> >> a=ptime:20 >>> >> a=sendrecv >>> >> >>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000@sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>> sofia/internal/10000@sip.domain.net entering state [calling][0] >>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000@sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000@sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>> 10000@sip.domain.net Update Callee ID to "10000" <10000> >>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>> sofia/internal/10000@sip.domain.net entering state [completing][200] >>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>> >> v=0 >>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>> >> s=pjmedia >>> >> c=IN IP4 10.0.0.131 >>> >> t=0 0 >>> >> a=X-nat:0 >>> >> m=audio 4010 RTP/AVP 9 101 >>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>> >> a=rtpmap:9 G722/8000 >>> >> a=sendrecv >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-15 >>> >> >>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000@sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>> signal sofia/internal/10000@sip.domain.net [BREAK] >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>> sofia/internal/10000@sip.domain.net entering state [ready][200] >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>> >> v=0 >>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>> >> s=pjmedia >>> >> c=IN IP4 10.0.0.131 >>> >> t=0 0 >>> >> a=X-nat:0 >>> >> m=audio 4010 RTP/AVP 9 101 >>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>> >> a=rtpmap:9 G722/8000 >>> >> a=sendrecv >>> >> a=rtpmap:101 telephone-event/8000 >>> >> a=fmtp:101 0-15 >>> >> >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>> G722 >>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>> send/recv payload to 101 >>> >> >>> >> >>> >> Thanks for any help, >>> >> Emrah >>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> >>> wrote: >>> >> >>> >>> What happened with the uuid_media_reneg command? >>> >>> -MC >>> >>> >>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: >>> >>> Hi all, >>> >>> >>> >>> How do we go about renegotiating the codecs of a call that is >>> already established? >>> >>> I tried uuid_media_reneg to no avail. >>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>> G722. >>> >>> >>> >>> Cheers and thanks, >>> >>> Emrah >>> >>> >>> _________________________________________________________________________ >>> >>> 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 >>> >>> >>> >>> 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 >>> >>> >>> >>> 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 >>> >> >>> >> 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 >>> > >>> > 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 >>> >>> 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 >> >> 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 > > 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
I do not have account on pastebin.freeswitch.org See: http://jira.freeswitch.org/browse/FS-5116 Thanks. 2013/2/20 Michael Collins <msc@freeswitch.org> > Definitely get the console log and SIP trace of those problem calls and > put them in pastein.freeswitch.org. > > -MC > > > On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com > > wrote: > >> Wow! This great feature is really work. I have the same issue - one way >> audio. And another: "show channels" always return first negotiated codec. >> When switching codec back, audio working fine. Tested with 3 different >> softphones. >> If someone interested, please confirm. >> Thanks. >> >> >> 2013/2/19 Michael Collins <msc@freeswitch.org> >> >>> pastebin the log. I'll bet Steven can diagnose it! ;) >>> -MC >>> >>> >>> On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists@kavun.ch> wrote: >>> >>>> Excellent, that worked. >>>> >>>> However I am having a one way audio issue, where the peer that has its >>>> codecs changed can hear but cannot be heard. >>>> Even if both bridged peers are changed to a common codec it doesn't >>>> work. >>>> >>>> Any idea? >>>> >>>> Thanks a bunch for your help. >>>> On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote: >>>> >>>> > Try: >>>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>>> > >>>> > (ie add = before PCMU) >>>> > >>>> > It appears to affect something in the code... >>>> > >>>> > -Steve >>>> > >>>> > >>>> > On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote: >>>> >> Hey, >>>> >> >>>> >> In short, nothing seems to happen. I tried several phones. >>>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>>> that the change in SDP with uuid_media_reneg would do it. >>>> >> >>>> >> This is what happens: >>>> >> freeswitch@internal> show channels >>>> >> >>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>>> 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah >>>> - Macbook >>>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k >>>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>>> cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, >>>> >> >>>> >> 1 total. >>>> >> >>>> >> freeswitch@internal> uuid_media_reneg >>>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>>> >> +OK Success >>>> >> >>>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>>> >> v=0 >>>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>>> >> s=FreeSWITCH >>>> >> c=IN IP4 1.2.3.4 >>>> >> t=0 0 >>>> >> m=audio 17888 RTP/AVP 9 0 101 >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-16 >>>> >> a=silenceSupp:off - - - - >>>> >> a=ptime:20 >>>> >> a=sendrecv >>>> >> >>>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>>> sofia/internal/10000@sip.domain.net entering state [calling][0] >>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>>> 10000@sip.domain.net Update Callee ID to "10000" <10000> >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>>> sofia/internal/10000@sip.domain.net entering state [completing][200] >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>>> >> v=0 >>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>> >> s=pjmedia >>>> >> c=IN IP4 10.0.0.131 >>>> >> t=0 0 >>>> >> a=X-nat:0 >>>> >> m=audio 4010 RTP/AVP 9 101 >>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>> >> a=rtpmap:9 G722/8000 >>>> >> a=sendrecv >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-15 >>>> >> >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>>> sofia/internal/10000@sip.domain.net entering state [ready][200] >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>>> >> v=0 >>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>> >> s=pjmedia >>>> >> c=IN IP4 10.0.0.131 >>>> >> t=0 0 >>>> >> a=X-nat:0 >>>> >> m=audio 4010 RTP/AVP 9 101 >>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>> >> a=rtpmap:9 G722/8000 >>>> >> a=sendrecv >>>> >> a=rtpmap:101 telephone-event/8000 >>>> >> a=fmtp:101 0-15 >>>> >> >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>>> G722 >>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>>> send/recv payload to 101 >>>> >> >>>> >> >>>> >> Thanks for any help, >>>> >> Emrah >>>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> >>>> wrote: >>>> >> >>>> >>> What happened with the uuid_media_reneg command? >>>> >>> -MC >>>> >>> >>>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: >>>> >>> Hi all, >>>> >>> >>>> >>> How do we go about renegotiating the codecs of a call that is >>>> already established? >>>> >>> I tried uuid_media_reneg to no avail. >>>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>>> G722. >>>> >>> >>>> >>> Cheers and thanks, >>>> >>> Emrah >>>> >>> >>>> _________________________________________________________________________ >>>> >>> 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 >>>> >>> >>>> >>> 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 >>>> >>> >>>> >>> 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 >>>> >> >>>> >> 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 >>>> > >>>> > 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 >>>> >>>> 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 >>> >>> 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 >> >> 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 > > 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 > >
Haha, you don't need an account. Re-read the challenge dialog on pastebin.freeswitch.org. -MC On Wed, Feb 20, 2013 at 11:57 AM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com>wrote: > I do not have account on pastebin.freeswitch.org > See: http://jira.freeswitch.org/browse/FS-5116 > Thanks. > > > 2013/2/20 Michael Collins <msc@freeswitch.org> > >> Definitely get the console log and SIP trace of those problem calls and >> put them in pastein.freeswitch.org. >> >> -MC >> >> >> On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko < >> dvl36.ripe.nick@gmail.com> wrote: >> >>> Wow! This great feature is really work. I have the same issue - one way >>> audio. And another: "show channels" always return first negotiated codec. >>> When switching codec back, audio working fine. Tested with 3 different >>> softphones. >>> If someone interested, please confirm. >>> Thanks. >>> >>> >>> 2013/2/19 Michael Collins <msc@freeswitch.org> >>> >>>> pastebin the log. I'll bet Steven can diagnose it! ;) >>>> -MC >>>> >>>> >>>> On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists@kavun.ch> wrote: >>>> >>>>> Excellent, that worked. >>>>> >>>>> However I am having a one way audio issue, where the peer that has its >>>>> codecs changed can hear but cannot be heard. >>>>> Even if both bridged peers are changed to a common codec it doesn't >>>>> work. >>>>> >>>>> Any idea? >>>>> >>>>> Thanks a bunch for your help. >>>>> On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote: >>>>> >>>>> > Try: >>>>> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU >>>>> > >>>>> > (ie add = before PCMU) >>>>> > >>>>> > It appears to affect something in the code... >>>>> > >>>>> > -Steve >>>>> > >>>>> > >>>>> > On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote: >>>>> >> Hey, >>>>> >> >>>>> >> In short, nothing seems to happen. I tried several phones. >>>>> >> I also tried enabling codec renegotiation on hold/unhold thinking >>>>> that the change in SDP with uuid_media_reneg would do it. >>>>> >> >>>>> >> This is what happens: >>>>> >> freeswitch@internal> show channels >>>>> >> >>>>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >>>>> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 >>>>> 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah >>>>> - Macbook >>>>> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k >>>>> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, >>>>> cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, >>>>> >> >>>>> >> 1 total. >>>>> >> >>>>> >> freeswitch@internal> uuid_media_reneg >>>>> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >>>>> >> +OK Success >>>>> >> >>>>> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >>>>> >> v=0 >>>>> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >>>>> >> s=FreeSWITCH >>>>> >> c=IN IP4 1.2.3.4 >>>>> >> t=0 0 >>>>> >> m=audio 17888 RTP/AVP 9 0 101 >>>>> >> a=rtpmap:101 telephone-event/8000 >>>>> >> a=fmtp:101 0-16 >>>>> >> a=silenceSupp:off - - - - >>>>> >> a=ptime:20 >>>>> >> a=sendrecv >>>>> >> >>>>> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel >>>>> sofia/internal/10000@sip.domain.net entering state [calling][0] >>>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ >>>>> 10000@sip.domain.net Update Callee ID to "10000" <10000> >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel >>>>> sofia/internal/10000@sip.domain.net entering state [completing][200] >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >>>>> >> v=0 >>>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>>> >> s=pjmedia >>>>> >> c=IN IP4 10.0.0.131 >>>>> >> t=0 0 >>>>> >> a=X-nat:0 >>>>> >> m=audio 4010 RTP/AVP 9 101 >>>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>>> >> a=rtpmap:9 G722/8000 >>>>> >> a=sendrecv >>>>> >> a=rtpmap:101 telephone-event/8000 >>>>> >> a=fmtp:101 0-15 >>>>> >> >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send >>>>> signal sofia/internal/10000@sip.domain.net [BREAK] >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel >>>>> sofia/internal/10000@sip.domain.net entering state [ready][200] >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >>>>> >> v=0 >>>>> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >>>>> >> s=pjmedia >>>>> >> c=IN IP4 10.0.0.131 >>>>> >> t=0 0 >>>>> >> a=X-nat:0 >>>>> >> m=audio 4010 RTP/AVP 9 101 >>>>> >> a=rtcp:4011 IN IP4 10.0.0.131 >>>>> >> a=rtpmap:9 G722/8000 >>>>> >> a=sendrecv >>>>> >> a=rtpmap:101 telephone-event/8000 >>>>> >> a=fmtp:101 0-15 >>>>> >> >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec >>>>> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using >>>>> G722 >>>>> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf >>>>> send/recv payload to 101 >>>>> >> >>>>> >> >>>>> >> Thanks for any help, >>>>> >> Emrah >>>>> >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> >>>>> wrote: >>>>> >> >>>>> >>> What happened with the uuid_media_reneg command? >>>>> >>> -MC >>>>> >>> >>>>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: >>>>> >>> Hi all, >>>>> >>> >>>>> >>> How do we go about renegotiating the codecs of a call that is >>>>> already established? >>>>> >>> I tried uuid_media_reneg to no avail. >>>>> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to >>>>> G722. >>>>> >>> >>>>> >>> Cheers and thanks, >>>>> >>> Emrah >>>>> >>> >>>>> _________________________________________________________________________ >>>>> >>> 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 >>>>> >>> >>>>> >>> 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 >>>>> >>> >>>>> >>> 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 >>>>> >> >>>>> >> 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 >>>>> > >>>>> > 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 >>>>> >>>>> 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 >>>> >>>> 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 >>> >>> 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 >> >> 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 > > 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
PB-20616 2013/2/20 Michael Collins <msc@freeswitch.org>
Haha, you don't need an account. Re-read the challenge dialog on pastebin.freeswitch.org.
-MC
On Wed, Feb 20, 2013 at 11:57 AM, Dmitry Lysenko < dvl36.ripe.nick@gmail.com> wrote:
I do not have account on pastebin.freeswitch.org See: http://jira.freeswitch.org/browse/FS-5116 Thanks.
2013/2/20 Michael Collins <msc@freeswitch.org>
Definitely get the console log and SIP trace of those problem calls and put them in pastein.freeswitch.org.
-MC
On Wed, Feb 20, 2013 at 5:04 AM, Dmitry Lysenko < dvl36.ripe.nick@gmail.com> wrote:
Wow! This great feature is really work. I have the same issue - one way audio. And another: "show channels" always return first negotiated codec. When switching codec back, audio working fine. Tested with 3 different softphones. If someone interested, please confirm. Thanks.
2013/2/19 Michael Collins <msc@freeswitch.org>
pastebin the log. I'll bet Steven can diagnose it! ;) -MC
On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists@kavun.ch> wrote:
Excellent, that worked.
However I am having a one way audio issue, where the peer that has its codecs changed can hear but cannot be heard. Even if both bridged peers are changed to a common codec it doesn't work.
Any idea?
Thanks a bunch for your help. On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre@gmail.com> wrote:
> Try: > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU > > (ie add = before PCMU) > > It appears to affect something in the code... > > -Steve > > > On 19 February 2013 05:54, Emrah <lists@kavun.ch> wrote: >> Hey, >> >> In short, nothing seems to happen. I tried several phones. >> I also tried enabling codec renegotiation on hold/unhold thinking that the change in SDP with uuid_media_reneg would do it. >> >> This is what happens: >> freeswitch@internal> show channels >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19 00:44:35,1361252675,sofia/internal/10000@sip.domain.net,CS_EXECUTE,Emrah - Macbook Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net@ek-custom-16k ,XML,ek-conference,G722,16000,64000,G722,16000,64000,, cornflakes.domain.net,10000@sip.domain.net,,ACTIVE,,,,,, >> >> 1 total. >> >> freeswitch@internal> uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU >> +OK Success >> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP: >> v=0 >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4 >> s=FreeSWITCH >> c=IN IP4 1.2.3.4 >> t=0 0 >> m=audio 17888 RTP/AVP 9 0 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:20 >> a=sendrecv >> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [calling][0] >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/ 10000@sip.domain.net Update Callee ID to "10000" <10000> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [completing][200] >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP: >> v=0 >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> s=pjmedia >> c=IN IP4 10.0.0.131 >> t=0 0 >> a=X-nat:0 >> m=audio 4010 RTP/AVP 9 101 >> a=rtcp:4011 IN IP4 10.0.0.131 >> a=rtpmap:9 G722/8000 >> a=sendrecv >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/10000@sip.domain.net [BREAK] >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/10000@sip.domain.net entering state [ready][200] >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP >> v=0 >> o=- 3570241475 3570241476 IN IP4 10.0.0.131 >> s=pjmedia >> c=IN IP4 10.0.0.131 >> t=0 0 >> a=X-nat:0 >> m=audio 4010 RTP/AVP 9 101 >> a=rtcp:4011 IN IP4 10.0.0.131 >> a=rtpmap:9 G722/8000 >> a=sendrecv >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722 >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 >> >> >> Thanks for any help, >> Emrah >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc@freeswitch.org> wrote: >> >>> What happened with the uuid_media_reneg command? >>> -MC >>> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists@kavun.ch> wrote: >>> Hi all, >>> >>> How do we go about renegotiating the codecs of a call that is already established? >>> I tried uuid_media_reneg to no avail. >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. >>> >>> Cheers and thanks, >>> Emrah >>> _________________________________________________________________________ >>> 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 >>> >>> 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 >>> >>> 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 >> >> 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 > > 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
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
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
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
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
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
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
Now this cool feature is fully working! Today Anthony committed the bugfix! Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. BTW, is there way to get API access to statistic of jitter buffer? Thanks. P.S. As I know, asterisk can't do such thing.
Hi guys, Sorry was disconnected from the list for a while. Indeed it looks like the latest commit fixed it. The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs. Great job and thanks a bunch, Emrah On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com> wrote:
Now this cool feature is fully working! Today Anthony committed the bugfix! Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. BTW, is there way to get API access to statistic of jitter buffer? Thanks.
P.S. As I know, asterisk can't do such thing. _________________________________________________________________________ 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
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
Freeswitch is turning into a mean, mean event-driven voip panacea... Congrats on the new commit. On Feb 22, 2013 6:49 AM, "Emrah" <lists@kavun.ch> wrote:
Hi guys,
Sorry was disconnected from the list for a while. Indeed it looks like the latest commit fixed it. The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs.
Great job and thanks a bunch, Emrah On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com> wrote:
Now this cool feature is fully working! Today Anthony committed the bugfix! Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. BTW, is there way to get API access to statistic of jitter buffer? Thanks.
P.S. As I know, asterisk can't do such thing. _________________________________________________________________________ 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
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
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
Is it necessary to file a bug on the channel info issue? The channel data does not get updated when changing codecs and a "show channel" always shows the first negotiated codec. This is an incredible feature and ever so useful, thanks a bunch. E On Feb 22, 2013, at 11:59 AM, Daniel Ivanov <sertys@gmail.com> wrote:
Freeswitch is turning into a mean, mean event-driven voip panacea... Congrats on the new commit.
On Feb 22, 2013 6:49 AM, "Emrah" <lists@kavun.ch> wrote: Hi guys,
Sorry was disconnected from the list for a while. Indeed it looks like the latest commit fixed it. The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs.
Great job and thanks a bunch, Emrah On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com> wrote:
Now this cool feature is fully working! Today Anthony committed the bugfix! Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. BTW, is there way to get API access to statistic of jitter buffer? Thanks.
P.S. As I know, asterisk can't do such thing. _________________________________________________________________________ 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
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
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
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
Emrah, please file a bug to jira. Btw, are you checked changing of codec ptime during call? Thanks. 2013/2/25 Emrah <lists@kavun.ch>
Is it necessary to file a bug on the channel info issue? The channel data does not get updated when changing codecs and a "show channel" always shows the first negotiated codec.
This is an incredible feature and ever so useful, thanks a bunch.
E On Feb 22, 2013, at 11:59 AM, Daniel Ivanov <sertys@gmail.com> wrote:
Freeswitch is turning into a mean, mean event-driven voip panacea... Congrats on the new commit.
On Feb 22, 2013 6:49 AM, "Emrah" <lists@kavun.ch> wrote: Hi guys,
Sorry was disconnected from the list for a while. Indeed it looks like the latest commit fixed it. The issue I see now is that the channel info are not updated and a show channels will only list the initial codecs.
Great job and thanks a bunch, Emrah On Feb 21, 2013, at 8:42 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com> wrote:
Now this cool feature is fully working! Today Anthony committed the bugfix! Tested with 2 different SIP UA<->FS<->Callcentric. Both legs. Now possible to write application that will switch codecs on the fly, regarding on any event, such as switching to low bandwidth backup internet line or too high cpu load, etc. BTW, is there way to get API access to statistic of jitter buffer? Thanks.
P.S. As I know, asterisk can't do such thing.
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
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
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
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
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
tsk tsk, gotta slow down and read. :) -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 20, 2013, at 3:01 PM, Dmitry Lysenko <dvl36.ripe.nick@gmail.com> wrote:
PB-20616
2013/2/20 Michael Collins <msc@freeswitch.org> Haha, you don't need an account. Re-read the challenge dialog on pastebin.freeswitch.org.
-MC
participants (6)
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Brian West
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Daniel Ivanov
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Dmitry Lysenko
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Emrah
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Michael Collins
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Steven Ayre