cannot make call, error INCOMPATIBLE_DESTINATION
Hi All, I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me. Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x [BREAK] (sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x [KILL] Send signal sofia/internal/963728839@x.x.x.x [BREAK] sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x) State HANGUP Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION Brs, Khue Nguyen.
Hello. There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs PCMU:0:8000:*5*:64000 PCMU:0:8000:*20*:64000 Antonio On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,
I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.
Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x [BREAK] (sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x [KILL] Send signal sofia/internal/963728839@x.x.x.x [BREAK] sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x) State HANGUP Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
Brs, Khue Nguyen.
_________________________________________________________________________ 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
Hi Antonio, How I can fix it? Khue. 2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com>
Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:*5*:64000
PCMU:0:8000:*20*:64000
Antonio
On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,
I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.
Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x [BREAK] (sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x [KILL] Send signal sofia/internal/963728839@x.x.x.x [BREAK] sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x) State HANGUP Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
Brs, Khue Nguyen.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
Tell your SIP client to use a ptime of 20ms On 6 June 2013 12:04, Khue Nguyen Minh <khuenm@vega.com.vn> wrote:
Hi Antonio,
How I can fix it?
Khue.
2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com>
Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:*5*:64000
PCMU:0:8000:*20*:64000
Antonio
On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,
I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.
Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x [BREAK] (sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x [KILL] Send signal sofia/internal/963728839@x.x.x.x [BREAK] sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x) State HANGUP Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
Brs, Khue Nguyen.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
Can I config ptime of FS to 5? 2013/6/6 Steven Ayre <steveayre@gmail.com>
Tell your SIP client to use a ptime of 20ms
On 6 June 2013 12:04, Khue Nguyen Minh <khuenm@vega.com.vn> wrote:
Hi Antonio,
How I can fix it?
Khue.
2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com>
Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:*5*:64000
PCMU:0:8000:*20*:64000
Antonio
On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,
I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.
Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x [BREAK] (sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x [KILL] Send signal sofia/internal/963728839@x.x.x.x [BREAK] sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x) State HANGUP Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
Brs, Khue Nguyen.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
Hello I have never seen ptime of 5. But you could try this : https://wiki.freeswitch.org/wiki/Config_vars.xml Or ( from my mind) originate {absolute_codec_string='PCMU@5i'}sofia/gateway/provdider/XXXXXXX But really your provider should be using ptime of 20 ( multiples of 10 ) you are just sending 4 X more packets this will probably take its toll on your infrastructure on the long run. Antonio On 6/6/13 12:57 PM, Khue Nguyen Minh wrote:
Can I config ptime of FS to 5?
2013/6/6 Steven Ayre <steveayre@gmail.com <mailto:steveayre@gmail.com>>
Tell your SIP client to use a ptime of 20ms
On 6 June 2013 12:04, Khue Nguyen Minh <khuenm@vega.com.vn <mailto:khuenm@vega.com.vn>> wrote:
Hi Antonio,
How I can fix it?
Khue.
2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com <mailto:eagle.antonio@gmail.com>>
Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:*5*:64000
PCMU:0:8000:*20*:64000
Antonio
On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,
I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.
Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> [BREAK] (sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x>) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> [KILL] Send signal sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> [BREAK] sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x <mailto:sofia/internal/963728839@x.x.x>.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x>) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x>) State HANGUP Channel sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x <mailto:sofia/internal/963728839@x.x.x.x> Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
Brs, Khue Nguyen.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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 <mailto: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
Maybe. But the ptime will have a minimum value and possibly need to be multiples of a value. The allowable ptimes are codec-dependant. 5ms may well be too low to be allowable. You can specify it using a @5i suffix in your codec preferences to override the default. But I would NOT recommend 5ms.... It refers to how long a sample to put in a single RTP frame (packet). Lower values mean you lose less audio data if you drop a packet but also mean you're sending more packets. Every packet has IP + UDP + RTP headers, so has a certain amount of overhead. There's a sweet spot between efficient bandwidth use and not losing too much data if packets are lost. With a 5ms ptime you're going to have a massive amount of overhead which isn't justified - the packets might well contain more header data than audio data. If you're on a slow connection you might even find you start getting packet loss you don't otherwise get because of the extra bandwidth being consumed by the headers. I would recommend changing the setting on the client instead to use 20ms. -Steve On 6 June 2013 12:57, Khue Nguyen Minh <khuenm@vega.com.vn> wrote:
Can I config ptime of FS to 5?
2013/6/6 Steven Ayre <steveayre@gmail.com>
Tell your SIP client to use a ptime of 20ms
On 6 June 2013 12:04, Khue Nguyen Minh <khuenm@vega.com.vn> wrote:
Hi Antonio,
How I can fix it?
Khue.
2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com>
Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:*5*:64000
PCMU:0:8000:*20*:64000
Antonio
On 6/6/13 11:12 AM, Khue Nguyen Minh wrote:
Hi All,
I'm using FS to receive call forward from GMSC. But, I have error INCOMPATIBLE_DESTINATION when new incoming call. This is log from fs_cli, please solve for me.
Audio Codec Compare [PCMA:8:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:5:64000]/[GSM:3:8000:20:13200] Audio Codec Compare [AMR-WB:101:8000:5:0]/[G722:9:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMU:0:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[PCMA:8:8000:20:64000] Audio Codec Compare [AMR-WB:101:8000:5:0]/[GSM:3:8000:20:13200] Audio Codec Compare [PCMU:0:8000:5:64000]/[G722:9:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMU:0:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMU:0:8000:5:64000]/[GSM:3:8000:20:13200] Substituting codec PCMU@20i@8000h No audio codec available No 2833 in SDP. Disable 2833 dtmf and switch to INFO Send signal sofia/internal/963728839@x.x.x.x [BREAK] (sofia/internal/963728839@x.x.x.x) Callstate Change RINGING -> HANGUP Hangup sofia/internal/963728839@x.x.x.x [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Send signal sofia/internal/963728839@x.x.x.x [KILL] Send signal sofia/internal/963728839@x.x.x.x [BREAK] sofia/internal/963728839@x.x.x.x skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) (sofia/internal/963728839@x.x.x.) State EXECUTE going to sleep (sofia/internal/963728839@x.x.x.x) Running State Change CS_HANGUP (sofia/internal/963728839@x.x.x.x) State HANGUP Channel sofia/internal/963728839@x.x.x.x hanging up, cause: INCOMPATIBLE_DESTINATION Responding to INVITE with: 488 sofia/internal/963728839@x.x.x.x Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
Brs, Khue Nguyen.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ 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
It looks like we only have implementations set in multiples of 10ms. You would have to modify code in switch_pcm.c in the loop where we add the implementations (around line 280). Mike On Jun 6, 2013, at 10:52 AM, Steven Ayre <steveayre@gmail.com> wrote:
Maybe. But the ptime will have a minimum value and possibly need to be multiples of a value. The allowable ptimes are codec-dependant. 5ms may well be too low to be allowable.
You can specify it using a @5i suffix in your codec preferences to override the default.
But I would NOT recommend 5ms.... It refers to how long a sample to put in a single RTP frame (packet). Lower values mean you lose less audio data if you drop a packet but also mean you're sending more packets. Every packet has IP + UDP + RTP headers, so has a certain amount of overhead. There's a sweet spot between efficient bandwidth use and not losing too much data if packets are lost. With a 5ms ptime you're going to have a massive amount of overhead which isn't justified - the packets might well contain more header data than audio data. If you're on a slow connection you might even find you start getting packet loss you don't otherwise get because of the extra bandwidth being consumed by the headers.
I would recommend changing the setting on the client instead to use 20ms.
-Steve
On 6 June 2013 12:57, Khue Nguyen Minh <khuenm@vega.com.vn> wrote: Can I config ptime of FS to 5?
2013/6/6 Steven Ayre <steveayre@gmail.com> Tell your SIP client to use a ptime of 20ms
On 6 June 2013 12:04, Khue Nguyen Minh <khuenm@vega.com.vn> wrote: Hi Antonio,
How I can fix it?
Khue.
2013/6/6 Antonio Teixeira <eagle.antonio@gmail.com> Hello.
There is a mismtach on the ptime http://wiki.freeswitch.org/wiki/Codecs
PCMU:0:8000:5:64000
PCMU:0:8000:20:64000
Antonio
participants (4)
-
Antonio Teixeira
-
Khue Nguyen Minh
-
Michael Jerris
-
Steven Ayre