INCOMPATIBLE_DESTINATION bridging GSM to mod_skypopen
I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers. When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase. 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated. Thanks, Danny Iland
mod_skypopen uses signed raw linear 16 bit (eg: SL16) natively, but you seem to force it to use PCMU or PCMA (eg: g711). FreeSWITCH will automatically do the conversion for you (from whatever your channel uses) when bridging to skypopen if you don't force it. -giovanni On Wed, Oct 16, 2013 at 8:04 PM, Danny Iland <iland@cs.ucsb.edu> wrote:
I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers.
When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase.
2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated.
Thanks, Danny Iland
_________________________________________________________________________ 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
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Giovanni, I know the Skype API takes a TCP stream of SL 16. In order to force the correct codecs, I inserted setting the following in the dialplan just before the bridge. <action application="export" data="nolocal:absolute_codec_string=L16,SL16,GSM"/> However call establishment still fails with INCOMPATIBLE_DESTINATION after comparing GSM and L16: 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[L16:70:8000:20:128000] Any suggestions on the contents of global_codec_prefs, outbound_codec_prefs, or nolocal:absolute_codec_string? I suspect I need to define something like L16@13200h but when I tried this, the L16 offering did not change. Full log snippet follows Thanks, Danny Iland 2013-10-17 18:22:25.252453 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype102/echo123] has been answered 2013-10-17 18:22:25.252453 [DEBUG] switch_channel.c:3594 (skypopen/skype102/echo123) Callstate Change RINGING -> ACTIVE 2013-10-17 18:22:25.252453 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype102 ][UP,INPROGRS] MSG_ID=41 2013-10-17 18:22:25.252453 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype102 ][UP,INPROGRS] outbound_channel_answered! 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[L16:70:8000:20:128000] 2013-10-17 18:22:25.272510 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-17 18:22:25.272510 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/[me]@[ip] [BREAK] 2013-10-17 18:22:25.272510 [NOTICE] switch_channel.c:3633 Hangup sofia/internal//[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-10-17 18:22:25.272510 [DEBUG] switch_channel.c:3134 Send signal sofia/internal//[me]@[ip] [KILL] 2013-10-17 18:22:25.272510 [DEBUG] switch_core_session.c:1338 Send signal sofia/internal//[me]@[ip] [BREAK] 2013-10-17 18:22:25.272510 [DEBUG] switch_ivr_originate.c:3450 sofia/internal//[me]@[ip] Media Establishment Failed. 2013-10-17 18:22:25.272510 [NOTICE] switch_ivr_originate.c:3452 Hangup skypopen/skype102/echo123 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] On Thu, Oct 17, 2013 at 1:00 AM, Giovanni Maruzzelli <gmaruzz@gmail.com>wrote:
mod_skypopen uses signed raw linear 16 bit (eg: SL16) natively, but you seem to force it to use PCMU or PCMA (eg: g711).
FreeSWITCH will automatically do the conversion for you (from whatever your channel uses) when bridging to skypopen if you don't force it.
-giovanni
On Wed, Oct 16, 2013 at 8:04 PM, Danny Iland <iland@cs.ucsb.edu> wrote:
I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers.
When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase.
2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated.
Thanks, Danny Iland
_________________________________________________________________________ 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
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_________________________________________________________________________ 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
On Oct 16, 2013 11:47 PM, "Danny Iland" <iland@cs.ucsb.edu> wrote:
I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers.
When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase.
2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated.
Thanks, Danny Iland
_________________________________________________________________________ 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
Solved on IRC - mod_spandsp was failing to load because of a build error where some libraries were missing. GSM wasn't available as a result, so FS couldn't initiate media once the call rang / was answered. http://xkcd.com/979/ On 17 October 2013 16:22, Aseem Anand <anandaseem89@gmail.com> wrote:
On Oct 16, 2013 11:47 PM, "Danny Iland" <iland@cs.ucsb.edu> wrote:
I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers.
When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase.
2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated.
Thanks, Danny Iland
_________________________________________________________________________ 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
:) On Thu, Oct 17, 2013 at 10:15 PM, Steven Ayre <steveayre@gmail.com> wrote:
Solved on IRC - mod_spandsp was failing to load because of a build error where some libraries were missing. GSM wasn't available as a result, so FS couldn't initiate media once the call rang / was answered.
On 17 October 2013 16:22, Aseem Anand <anandaseem89@gmail.com> wrote:
On Oct 16, 2013 11:47 PM, "Danny Iland" <iland@cs.ucsb.edu> wrote:
I'm using Freeswitch as a PBX for OpenBTS, to route calls from mobile phones to VoIP providers.
When bridging a call from a cellular handset to a mod_skypopen interface, I get the following error as soon as the Skype call is answered. The handset then hangs up. It appears the problem lies in the 'audio codec compare' phase.
2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3548 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.932473 [NOTICE] mod_skypopen.c:2471 Channel [skypopen/skype101/echo123] has been answered 2013-10-16 17:54:21.932473 [DEBUG] switch_channel.c:3594 (skypopen/skype101/echo123) Callstate Change RINGING -> ACTIVE 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:1232 [543dc3c|22dd4bf] [DEBUG_SKYPE 1232 ][skype101 ][UP,INPROGRS] MSG_ID=41 2013-10-16 17:54:21.932473 [DEBUG] mod_skypopen.c:2479 [543dc3c|22dd4bf] [DEBUG_SKYPE 2479 ][skype101 ][UP,INPROGRS] outbound_channel_answered! 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1172 [543dc3c|22dd4bf] [DEBUG_SKYPE 1172 ][skype101 ][UP,INPROGRS] ACCEPTED here you send me 32776 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1177 [543dc3c|22dd4bf] [DEBUG_SKYPE 1177 ][skype101 ][UP,INPROGRS] 4 SO_RCVBUF is 5120, size is 4 2013-10-16 17:54:21.932473 [DEBUG] skypopen_protocol.c:1181 [543dc3c|22dd4bf] [DEBUG_SKYPE 1181 ][skype101 ][UP,INPROGRS] 4 SO_SNDBUF is 5120, size is 4 2013-10-16 17:54:21.952511 [DEBUG] skypopen_protocol.c:207 [543dc3c|22dd4bf] [DEBUG_SKYPE 207 ][skype101 ][UP,INPROGRS] READING: |||ALTER CALL 32 SET_INPUT PORT="32776"||| 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5226 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2013-10-16 17:54:21.952511 [DEBUG] sofia_glue.c:5370 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-10-16 17:54:21.952511 [DEBUG] switch_core_session.c:858 Send signal sofia/internal/IMSI[me]@[ip] [BREAK] 2013-10-16 17:54:21.952511 [NOTICE] switch_channel.c:3633 Hangup sofia/internal/IMSI[me]@[ip] [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Any advice on forcing a transcode or otherwise having the Audio Codec Compare result in a successful bridge is appreciated.
Thanks, Danny Iland
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ 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
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
participants (4)
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Aseem Anand
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Danny Iland
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Giovanni Maruzzelli
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Steven Ayre