Re: [asterisk-users] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Rainer Piper wrote: perhaps a silly question ... if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ? if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ? The bridge_native_rtp module can actually native bridge in two ways: 1. Media directly between both sides 2. Media within the RTP stack Even with NAT #2 can still operate fine as media still goes through Asterisk, just not as much. As for your issue I would suggest you get a complete console log output with debug and create an issue[1] as this sounds like a bug. Cheers, [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place Thanks! Rainer Am 07.05.2014 12:36, schrieb Joshua Colp: Rainer Piper wrote: perhaps a silly question ... if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ? if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ? The bridge_native_rtp module can actually native bridge in two ways: 1. Media directly between both sides 2. Media within the RTP stack Even with NAT #2 can still operate fine as media still goes through Asterisk, just not as much. As for your issue I would suggest you get a complete console log output with debug and create an issue[1] as this sounds like a bug. Cheers, [1] https://issues.asterisk.org/jira -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Rainer Piper wrote: Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not t early. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
and I get ready for launch in germany at 13:15 ;-) Am 07.05.2014 13:09, schrieb Joshua Colp: Rainer Piper wrote: Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not t early. -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
upps ... off topic and typo lunch not launch ;-) Am 07.05.2014 13:14, schrieb Rainer Piper: and I get ready for launch in germany at 13:15 ;-) Am 07.05.2014 13:09, schrieb Joshua Colp: Rainer Piper wrote: Hi Joshua, I'll give it a try at ttps://issues.asterisk.org/jira and I hope my English is good enough to explain the problem. ;-) wow ... early bird it is 03:36 (PDT) in the morning at your place The office is in Alabama so it is 6:09AM there. I'm in Atlantic Canada though where it is 8:09AM. Not t early. -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY server*CLI core show channel PJSIP/7000-0001 -- General -- Name: PJSIP/7000-0001 Type: PJSIP UniqueID: 1399382022.1 LinkedID: 1399382022.0 Caller ID: 7000 Caller ID Name: (N/A) Connected Line ID: 7001 Connected Line ID Name: 7001 Eff. Connected Line ID: 7001 Eff. Connected Line ID Name: 7001 DNID Digits: (N/A) Language: de State: Up (6) NativeFormats: (alaw) WriteFormat: g722 ReadFormat: g722 WriteTranscode: Yes (g722)-(slin)-(alaw) ReadTranscode: Yes (alaw)-(slin)-(g722) Time to Hangup: 0 Elapsed Time: 0h3m24s Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension:pjsi Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Call Identifer: [C-] Variables: BRIDGEPEER=PJSIP/7001- DIALEDPEERNUMBER=7000 CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid= 700 level 1: src=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: lastapp=AppDial level 1: lastdata=(Outgoi level 1: start=1399382 level 1: answer=1399382 level 1: end=1399382 level 1: duration=1 level 1: billsec=0 level 1: disposition=8 level 1: amaflags=3 level 1: uniqueid=1399382 level 1: linkedid=1399382 level 1: sequence=1 server*CLI core show channel PJSIP/7001- -- General -- Name: PJSIP/7001- Type: PJSIP UniqueID: 1399382022.0 LinkedID: 1399382022.0 Caller ID: 7001 Caller ID Name: 7001 Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: de State: Up (6) NativeFormats: (g722) WriteFormat: g722 ReadFormat: g722 WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m51s Bridge ID: 0cbdb7d7-81ed-4c5f-896f-cea37599b148 -- PBX -- Context: outgoing-kamailio Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: Dial Data: PJSIP/7000 Call Identifer: [C-] Variables: BRIDGEPEER=PJSIP/7000-0001 DIALEDPEERNUMBER=7000 DIALEDPEERNAME=PJSIP/7000-0001 DIALSTATUS=ANSWER DIALEDTIME= ANSWEREDTIME= CDR Variables: level 1: calledsubaddr= level 1: callingsubaddr= level 1: dnid= level 1: clid=7001 level 1: src=7001 level 1: dst=7000 level 1: dcontext=outgoin level 1: channel=PJSIP/7 level 1: dstchannel=PJSIP/7 level 1: lastapp=Dial level 1: lastdata=PJSIP/7 level 1: start=1399382 level 1: answer=1399382 level 1: end=0.0 level 1: duration=230 level 1: billsec=228 level 1: disposition=8 level 1: amaflags=3 level 1: uniqueid=1399382 level 1: linkedid=1399382 level 1: sequence=0 -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
that's funny I recompiled asterisk without bridge_native_rtp.so to force asterisk to go to simple_bridge and not to native_bridge... !!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu Am 07.05.2014 07:11, schrieb Rainer Piper: PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de NOC +49 228 97167161 - sip.soho-piper.de NOC +882 990111550 via e164.org International Network NOC +49 2247 9064188 - sip.tefonix.de - D293 NOC +882 990045450 via e164.org International Network -- _ -- 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] asterisk12.2.0 PJSIP2.2.0 codec translation problem
perhaps a silly question ... if a channel switches from simple_bridge to native_bridge ... is the channel switching to direct_media between the endpoints ? if so, why doesn't turn direct_media = no and disable_direct_media_on_nat = yes switching to native_bridge off ? my pjsip.conf endpoint 7000 and 7001 [7000] type=endpoint context=outgoing disallow=all allow=alaw,ulaw,g722 transport=transport-udp auth=auth7000 aors=7000 direct_media = no disable_direct_media_on_nat = yes [auth7000] type=auth auth_type=userpass password=x username=7000 [7000] type=aor max_contacts=10 qualify_frequency=60 [7001] type=endpoint context=outgoing disallow=all allow=g722,alaw,ulaw transport=transport-udp auth=auth7001 aors=7001 direct_media = no disable_direct_media_on_nat = yes [auth7001] type=auth auth_type=userpass password=x username=7001 [7001] type=aor max_contacts=10 qualify_frequency=60 Am 07.05.2014 07:35, schrieb Rainer Piper: that's funny I recompiled asterisk without bridge_native_rtp.so to force asterisk to go to simple_bridge and not to native_bridge... !!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu Am 07.05.2014 07:11, schrieb Rainer Piper: PS. if I configure both extension 7000 and 7001 to, disallow=all allow=alaw or disallow=all allow=g722 everything is fine. as long as the allowed codec is equal in both extensions. Am 07.05.2014 07:00, schrieb Rainer Piper: Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY www.soho-piper.de http://www.soho-piper.de -- -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- 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
