[asterisk-users] Codec Mismatch
Sometimes in huge call volume am facing this type of error, [Jun 4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write: Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:04] WARNING[8285][C-79da]: channel.c:5075 ast_write: Codec mismatch on channel Local/6513@xss-call-out-4775;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:10] WARNING[8790][C-7a2c]: channel.c:5075 ast_write: Codec mismatch on channel Local/18002662279@xss-call-out-4778;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:23] WARNING[8355][C-79e6]: channel.c:5075 ast_write: Codec mismatch on channel Local/2896@xss-call-out-4779;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:25] WARNING[7577][C-798a]: channel.c:5075 ast_write: Codec mismatch on channel Local/2896@xss-call-out-477a;1 setting write format to slin from ulaw native formats (ulaw) basically Asterisk will do the slin to ulaw, hope there should not be any problem... But am not sure why am getting this error? will this affect my call? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec mismatch on channel
I get this on 1.8.x as well. I assume it is a harmless bug. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot Sent: Wednesday, February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec mismatch on channel
I think it's a warning as opposed to a bug. If the call were happening all in Tecnology (SIP/DAHDI/etc), the warning would be because your channel didn't support the codec (I can't do alaw so I'm gonna talk in slin). The LOCAL channel by definition (AFAIK) doesn't specifically support/deny any codec format. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, February 22, 2012 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] codec mismatch on channel I get this on 1.8.x as well. I assume it is a harmless bug. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot Sent: Wednesday, February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec mismatch on channel
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, February 22, 2012 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] codec mismatch on channel I get this on 1.8.x as well. I assume it is a harmless bug. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Nowrot Sent: Wednesday, February 22, 2012 3:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] codec mismatch on channel Hi I am keep getting this warning message when doing attendant transfer: WARNING[6027] channel.c: Codec mismatch on channel Local/XX@Inside-1f32;1 setting write format to slin from alaw native formats (alaw) What can I do to lose it. I am using asterisk 10.1.2 Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
Hi, I have strange problem. I have 1 SIP client (bt100) and 1 Iax2 client (IAXPhone): - when I call from Iax to SIP sound works - when I call from Sip to Iax sound doesn't work, I get : Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Any advice, help ? Thanks in advance, regards, Rob. In both configs there are only general codec settings . I have in sip.conf (snippet): [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm context = from-sip ; Send unknown SIP callers to this context And in iax.conf (snippet) : [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) ;delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=yes mailboxdetail=yes authdebug=yes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
(IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
- Original Message - From: Steve Kann [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 3:56 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone Robert Rozman wrote: - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone (IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm Oh, I see my mistake. Asterisk checks settings in iax.conf and plays accordingly (I thought it talks to client by himself). I think it should, but the old version of iaxclient that iaxphone is based on doesn't advertise it's supported codecs correctly.. -SteveK Hi, thanks for info. Which iax softphones are using newer iaxclient ? What is the best iax softphone from this point of view ? REgards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
Robert Rozman wrote: - Original Message - From: Steve Kann [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 3:56 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone Robert Rozman wrote: - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone (IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm Oh, I see my mistake. Asterisk checks settings in iax.conf and plays accordingly (I thought it talks to client by himself). I think it should, but the old version of iaxclient that iaxphone is based on doesn't advertise it's supported codecs correctly.. -SteveK Hi, thanks for info. Which iax softphones are using newer iaxclient ? What is the best iax softphone from this point of view ? I don't know for sure, but I think iaxcomm and DIAX are most up-to-date. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
Robert Rozman wrote: - Original Message - From: [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone (IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm Oh, I see my mistake. Asterisk checks settings in iax.conf and plays accordingly (I thought it talks to client by himself). I think it should, but the old version of iaxclient that iaxphone is based on doesn't advertise it's supported codecs correctly.. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
- Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 2:44 PM Subject: Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone (IAXPhone): I suppose you're talking about Steve Sokol's phone If so, then this phone only support gsm. Jan 25 13:52:22 NOTICE[31334]: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/200/1 of format gsm since our native format has changed to ulaw Why is Asterisk not satisfied with gsm packets - it should transcode if necessary ? I have enabled gsm and ulaw in both configs, but it seems not sufficient. Yes, * will transcode, but you specified in the IAX Phone config that you allow this one tu use gsm AND ulaw, so instead of transcoding, * just tell the IAX Phone to switch to uLaw, since the originating party sends it in ulaw. Just change your iax.conf to only allow gsm on the IAX Phone like this : disallow=all allow=gsm Oh, I see my mistake. Asterisk checks settings in iax.conf and plays accordingly (I thought it talks to client by himself). Thanks for help, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec mismatch between SIP (BT) and IAX Phone
thanks for info. Which iax softphones are using newer iaxclient ? What is the best iax softphone from this point of view ? I don't know for sure, but I think iaxcomm and DIAX are most up-to-date. I'm almost finished building my IAX softphone that is based on a recent version of iaxclient (one from the beginning of this month). I't will be free for non-commercial use. I'll post it here when it's available ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
