[asterisk-users] early media (video)
Hi All, I've been looking for information on how to use asterisk and early media to allow for a video-preview of the caller at the callee's phone for days... but I haven't been too successful :( I found that there seems to be a company 2N Helios IP which claims (youtube-video) that their SIP server is able to provide early video (using a Grandstream 3157v2 with preview enabled), but I would like to have this with asterisk... I'm currently using asterisk 12.2.x. I tried with all kinds of combinations of prematuremedia and progressinband in sip.conf and many different dialplan-extension-scripts but to no avail... sniffing with wireshark shows me, that the caller (doorstation) is sending H.264 video but the RTP video stream is not passed on to the callee by asterisk. (establishing a direct-video - without preview - call does work of course) is it at all possible with a default asterisk installation? maybe using chan_pjsip instead of chan_sip? do the prematuremedia and progressinband properties only apply to audio and not to video? I think this feature is essential for a sip-device which is used as a door-station... any information / idea is highly appreciated! thanks, Fronc -- _ -- 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] early media (video)
FYI: Joshua Colp already replied to my initial post of this message in asterisk-app-dev. he suggested to move it here (asterisk-users) he so far stated, that early media/Video should theoretically work... but probably no one tried this in recent times... looking foreward to receive further information, thanks! On Wed, May 7, 2014 at 9:09 AM, Fronc Hias fronc.h...@gmail.com wrote: Hi All, I've been looking for information on how to use asterisk and early media to allow for a video-preview of the caller at the callee's phone for days... but I haven't been too successful :( I found that there seems to be a company 2N Helios IP which claims (youtube-video) that their SIP server is able to provide early video (using a Grandstream 3157v2 with preview enabled), but I would like to have this with asterisk... I'm currently using asterisk 12.2.x. I tried with all kinds of combinations of prematuremedia and progressinband in sip.conf and many different dialplan-extension-scripts but to no avail... sniffing with wireshark shows me, that the caller (doorstation) is sending H.264 video but the RTP video stream is not passed on to the callee by asterisk. (establishing a direct-video - without preview - call does work of course) is it at all possible with a default asterisk installation? maybe using chan_pjsip instead of chan_sip? do the prematuremedia and progressinband properties only apply to audio and not to video? I think this feature is essential for a sip-device which is used as a door-station... any information / idea is highly appreciated! thanks, Fronc -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 07/05/2014 00:57, Rusty Newton a écrit : On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote: snip As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): snip NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. despite the fact that modification was done in a .conf file. I took this example as with module reload app_queue the above message appears. For sip, iax, voicemail, aso there is no message, just SIP reload or ... To make asterisk take the modification in account, you have to open /etc/asterisk/[sip|iax|voicemail|queue|..].conf and save it without making any change. After this the command will be execute. It you run it a second time in a raw, you will see that the false behavior appears again till you again open/save the original file. Hi! Hello I tried to reproduce using your description here and could not reproduce the issue. I tried with both sip.conf and queues.conf. Making a change in an included .conf file, but NOT the parent .conf file and then reloading that module from the CLI results in: centosclean*CLI module reload app_queue.so -- Reloading module 'app_queue.so' (True Call Queueing) [May 6 17:51:39] NOTICE[16211]: app_queue.c:7765 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. == Parsing '/etc/asterisk/queues.conf': Found == Parsing '/etc/asterisk/queue_include_1.conf': Found == Parsing '/tmp/queue_include_2.conf': Found I get the same behavior with sip.conf, it appears to work fine, whether I'm making only changes in the parent .conf or the included children. I even tried with two different included files in each sip.conf and queues.conf, one in /tmp and one in /etc/asterisk. Same working behavior. Ok, let's explain our files conf. /etc/asterisk/all asterisk original conf files /etc/asterisk/local/[additional_sip-general|additional_iax-general|...].conf /etc/asterisk/local/[extensions.d|sip.d|iax.d|queues.d|voicemail.d|...]/local conf files.conf The local directory and all his subdirectories are owned by a normal user with 755 rights Configuration set sample with sip.conf: In /etc/asterisk/sip.conf, we have #include local/additionnal_sip-general.conf in one place in the file and at the end of the file we have #include local/sip.d/*.conf This setup is the same since ages and was working well. Remember that switching back to previous version 1.8.26.1 or 11.8.1 make this setup working again. We have this proplem on all servers we upgraded to last asterisk version I used SVN-branch-11-r413305, so you might want to test there. However I'm still confused as to how you are seeing the behavior you are seeing. Servers are in production. Thanks for your support -- Daniel -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 07/05/2014 01:28, Steve Edwards a écrit : On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI ad...@tootai.net wrote: snip As explained in one on my previous message, it's a bug, easily reproducible: take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like this (what is important is the #include): snip NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. On Tue, 6 May 2014, Rusty Newton wrote: However I'm still confused as to how you are seeing the behavior you are seeing. Any chance the OP is including files from a file system that isn't maintaining atime/ctime/mtime/etc as expected, like NFS? No :-) See my previous answer. -- Daniel -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 07/05/2014 00:57, Rusty Newton a écrit : [...] I tried to reproduce using your description here and could not reproduce the issue. I tried with both sip.conf and queues.conf. Making a change in an included .conf file, but NOT the parent .conf file and then reloading that module from the CLI results in: centosclean*CLI module reload app_queue.so -- Reloading module 'app_queue.so' (True Call Queueing) [May 6 17:51:39] NOTICE[16211]: app_queue.c:7765 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. == Parsing '/etc/asterisk/queues.conf': Found == Parsing '/etc/asterisk/queue_include_1.conf': Found == Parsing '/tmp/queue_include_2.conf': Found I get the same behavior with sip.conf, it appears to work fine, whether I'm making only changes in the parent .conf or the included children. I even tried with two different included files in each sip.conf and queues.conf, one in /tmp and one in /etc/asterisk. Same working behavior. I got it: if the filename is given in totality it's working (as you do it). It's the #include /path to directorie/*.conf which is not taking in account (here *.conf description) -- Daniel -- _ -- 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: 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] early media (video)
Fronc Hias wrote: FYI: Joshua Colp already replied to my initial post of this message in asterisk-app-dev. he suggested to move it here (asterisk-users) he so far stated, that early media/Video should theoretically work... but probably no one tried this in recent times... looking foreward to receive further information, thanks! If you can provide the console output with debug and dialplan I may be able to assist further. It would also be useful to try Asterisk 11 and see if this is a regression. Cheers, -- 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] Video with asterisk12 and pjsip
Hi, I tried to turn on Video and get the following cli-WARNING output -- Executing [8600@outgoing-kamailio:1] Answer(PJSIP/7000-, ) in new stack 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 -- Executing [8600@outgoing-kamailio:2] ConfBridge(PJSIP/7000-, 8600) in new stack -- PJSIP/7000- Playing 'conf-onlyperson.g722' (language 'de') -- PJSIP/7000- Playing 'confbridge-join.g722' (language 'de') -- CBAnn/8600-;1 Playing 'confbridge-join.slin' (language 'en') -- Channel CBAnn/8600-;2 joined 'softmix' base-bridge 52997aa1-eb00-481c-8c56-e26d78d01515 -- Channel CBAnn/8600-;2 left 'softmix' base-bridge 52997aa1-eb00-481c-8c56-e26d78d01515 -- Started music on hold, class 'default', on channel 'PJSIP/7000-' -- Channel PJSIP/7000- joined 'softmix' base-bridge 52997aa1-eb00-481c-8c56-e26d78d01515 [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats [May 7 16:21:32] WARNING[20789]: channel.c:834 ast_best_codec: Don't know any of (h263|h263p|h264) formats 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 0x7f46f4187280 -- Probation passed - setting RTP source address to 192.168.8.203:31384 Endpoint 7000 is a Grandstream GXV3175 with Video the pjsip.conf for exten 7000 is [7000] type=endpoint context=outgoing-kamailio disallow=all allow=g722,alaw,ulaw,h264,h263p,h263,h261 transport=transport-udp auth=auth7000 aors=7000 direct_media=no disable_direct_media_on_nat=yes do I have to turn on the Video Support somewhere else ? -- *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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote: I got it: if the filename is given in totality it's working (as you do it). It's the #include /path to directorie/*.conf which is not taking in account (here *.conf description) That still works for me as well. I switched to Asterisk 11.9.0 built fresh from a tarball with default compilation options. I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf I then edited /etc/asterisk/sip_includes/sip_included.conf to add a SIP peer. Started Asterisk. Then edited only the sip_included.conf file to change the peer name. Connected to Asterisk console, performed 'sip show peers', 'sip reload', 'sip show peers'. Everything worked as expected: here is a pastebin: http://pastebin.com/XGKKu4x9 That is, when using a wildcard in the file path in an include inside sip.conf, Asterisk correctly detects a change in the included conf file upon a sip reload. You might try reproducing the issue on a fresh install, on your non-production system to see if you can narrow down where the difference is. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 07/05/2014 16:50, Rusty Newton a écrit : On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote: I got it: if the filename is given in totality it's working (as you do it). It's the #include /path to directorie/*.conf which is not taking in account (here *.conf description) That still works for me as well. I switched to Asterisk 11.9.0 built fresh from a tarball with default compilation options. I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf Here is the point. Modify it the way explained in previous message, like #include sip_includes/*.conf You should face the problem. And if you run it twice in a raw, it will do nothing the second time. I then edited /etc/asterisk/sip_includes/sip_included.conf to add a SIP peer. Started Asterisk. Then edited only the sip_included.conf file to change the peer name. Connected to Asterisk console, performed 'sip show peers', 'sip reload', 'sip show peers'. sip show peers was always working, only reload (remember, not only sip stuff) did nothing. Everything worked as expected: here is a pastebin: http://pastebin.com/XGKKu4x9 That is, when using a wildcard in the file path in an include inside sip.conf, Asterisk correctly detects a change in the included conf file upon a sip reload. You might try reproducing the issue on a fresh install, on your non-production system to see if you can narrow down where the difference is. I reinstall our test server from scratch before opening the bug. As I also told, reinstalling previous versions solve the problem. -- Daniel -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Administrator TOOTAI wrote: Le 07/05/2014 16:50, Rusty Newton a écrit : On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote: I got it: if the filename is given in totality it's working (as you do it). It's the #include /path to directorie/*.conf which is not taking in account (here *.conf description) That still works for me as well. I switched to Asterisk 11.9.0 built fresh from a tarball with default compilation options. I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf Here is the point. Modify it the way explained in previous message, like #include sip_includes/*.conf You should face the problem. And if you run it twice in a raw, it will do nothing the second time. Can you clarify this specific point? It's actually expected that if nothing changes and you do a reload that nothing will happen. Just so we're on the same page here... #1 Do you mean that you make a change, do a reload, and nothing happens. OR #2 That you make a change, do two reloads, and the second one does nothing. I was under the impression that #1 was going on. -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 07/05/2014 17:22, Joshua Colp a écrit : Administrator TOOTAI wrote: Le 07/05/2014 16:50, Rusty Newton a écrit : On Wed, May 7, 2014 at 2:46 AM, Administrator TOOTAI ad...@tootai.net wrote: I got it: if the filename is given in totality it's working (as you do it). It's the #include /path to directorie/*.conf which is not taking in account (here *.conf description) That still works for me as well. I switched to Asterisk 11.9.0 built fresh from a tarball with default compilation options. I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf Here is the point. Modify it the way explained in previous message, like #include sip_includes/*.conf You should face the problem. And if you run it twice in a raw, it will do nothing the second time. Can you clarify this specific point? It's actually expected that if nothing changes and you do a reload that nothing will happen. Just so we're on the same page here... It's 1.8.27 and 11.9.0 specific? I have in mind that if you do a reload even without any change the reload will going on. Anyway, it make sense to do nothing when there is no change. Please forget the if you run it twice in a raw I wrote above. #1 Do you mean that you make a change, do a reload, and nothing happens. OR #2 That you make a change, do two reloads, and the second one does nothing. I was under the impression that #1 was going on. #1 is the result of the bug related to the #include path #2 you explain me that it's expected behavior, I don't have to focuse on it -- Daniel -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, May 7, 2014 at 10:14 AM, Administrator TOOTAI ad...@tootai.net wrote: Le 07/05/2014 16:50, Rusty Newton a écrit : I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf Here is the point. Modify it the way explained in previous message, like #include sip_includes/*.conf You should face the problem. And if you run it twice in a raw, it will do nothing the second time. Unfortunately, no. I went ahead and tried this as well. I still get working behavior even when using #include sip_includes/*.conf -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
Le 07/05/2014 18:53, Rusty Newton a écrit : On Wed, May 7, 2014 at 10:14 AM, Administrator TOOTAI ad...@tootai.net wrote: Le 07/05/2014 16:50, Rusty Newton a écrit : I contructed a basic sip.conf, and added this line to the end: #include /etc/asterisk/sip_includes/*.conf Here is the point. Modify it the way explained in previous message, like #include sip_includes/*.conf You should face the problem. And if you run it twice in a raw, it will do nothing the second time. Unfortunately, no. I went ahead and tried this as well. I still get working behavior even when using #include sip_includes/*.conf Please try the includes *exactly* as I have them in sip.conf (same directories name and subdirectories) knowing that local is in /etc/asterisk sip.conf [general] context=default-SIP ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) realm=sip2.tootai.net ; Realm for digest authentication udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) transport=udp srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all ; First disallow all codecs allow=g722 ; Allow codecs in order of preference allow=ulaw allow=alaw allow=h264 allow=h263p allow=h263 language=fr ; Default language setting for all users/peers useragent=TOOTAiAudio ; Allows you to change the user agent string sdpsession=TOOTAiAudio PBX videosupport=yes; Turn on support for SIP video. You need to turn this alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, registerattempts=0 ; try for ever (default=10) registertimeout=20 ; default #include local/additional_sip-general.conf #include local/additional_sip-register.conf [authentication] #include local/sip.d/*.conf -- Daniel -- _ -- 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] Reload problems with 1.8.27 and 11.9.0 - Someone else ?
On Wed, May 7, 2014 at 12:59 PM, Administrator TOOTAI ad...@tootai.net wrote: Please try the includes *exactly* as I have them in sip.conf (same directories name and subdirectories) knowing that local is in /etc/asterisk I used your identical config to narrow it down. I re-opened https://issues.asterisk.org/jira/browse/ASTERISK-23683 and edited the Summary and Description fields, as well as linked it an issue where the fix for that issue *may* have introduced the problem you found. Thanks! -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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
[asterisk-users] Ghost calls on PBX
Hi all, I have a user with an old Mitel PBX connected to a couple of SPA112's. The user is reporting that their phones ring several times a day and when they answer the call, all they hear is dial tone or busy signal. Their PBX guy says that the SPA112's aren't providing line supervision and the PBX requires it. Does anyone know how to fix this? I'd also like to fix it from a provisioning file, if possible. Thank you! Mike. -- _ -- 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] les opérations ajouter,supprimer,modifier un client avec jEE et asterisk
salut, je suis entrain de developper une application jEE avec asterisk qui consiste à ajouter, supprimer,modifier des clients en utilisant Asterisk au lieu de base de donnée, donc je suis débutante dans ce domaine,j'ai fait pour l'instant une connection via asterisk avec API manager , mais je n'arrive pas a faire la manipulation d'ajout,supprime..Client car je n'ai pas saisie le client a modifier dans Asterisk c'est quoi, le usr ou peer ou quoi? et en API Asterisk en java je ne trouve pas des méthodes prédéfinies qui me permettent d'arriver à mes fins. Autre question:l'approche d'utiliser les sockets est-elle possible pour réaliser le CRUD de Client ? une aide me serait bien utile. -- _ -- 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] Ghost calls on PBX
Most FXS ATAs do not support supervision so they don't work well when plugged into a PBX's analog FXO (aka CO) ports. If the Mitel can provide supervision on analog phone ports (i.e. FSX) then you could use an ATA with FXO ports. If the Mitel does not support supervision on analog phone ports (FXS) this won't work either. You could also avoid all the hassle, stress, and heartache from trying to stick ATAs on a non-VoIP PBX and replace the Mitel. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl Sent: Wednesday, May 07, 2014 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Ghost calls on PBX Hi all, I have a user with an old Mitel PBX connected to a couple of SPA112's. The user is reporting that their phones ring several times a day and when they answer the call, all they hear is dial tone or busy signal. Their PBX guy says that the SPA112's aren't providing line supervision and the PBX requires it. Does anyone know how to fix this? I'd also like to fix it from a provisioning file, if possible. Thank you! Mike. -- _ -- 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] CBAnn channel not going away in Asterisk 12
On Tue, May 6, 2014 at 1:01 PM, Richard Kenner ken...@gnat.com wrote: That is definitely a leak and the fix looks good. Thanks. That leak is most likely the one biting you. It definitely is. Committed the fix for this leak on Asterisk v12 branch in -r413454. There is another leak in handle_cli_confbridge_kick() if the participant to kick is not in the conference. Confirmed. I missed that one in my code reading. I just fixed it the same way. Committed the fix for this leak on Asterisk v12 branch in -r413452. This leak also applied to Asterisk v11. Richard -- _ -- 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] pulseaudio question and console
If I login as a user and run asterisk it connects to pulse and runs. if I login as root and run the command su user -c asterisk -vc it does not connect to pulse and run. I thought su ran as that user - am I missing something to get the su command to run correctly and connect to pulseaudio as that user. Thanks, Jerry -- _ -- 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] les opérations ajouter,supprimer,modifier un client avec jEE et asterisk
On Wed, 7 May 2014, Meriem Abid wrote: salut, je suis entrain de developper une application... You will have better luck if you can post in English. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] CBAnn channel not going away in Asterisk 12
Committed the fix for this leak on Asterisk v12 branch in -r413452. This leak also applied to Asterisk v11. Thanks. Is this for both the one in the talking callback or the one in handle_cli_confbridge_kick or both (the fix is similar in both)? -- _ -- 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] CBAnn channel not going away in Asterisk 12
On Wed, May 7, 2014 at 4:43 PM, Richard Kenner ken...@gnat.com wrote: Committed the fix for this leak on Asterisk v12 branch in -r413452. This leak also applied to Asterisk v11. Thanks. Is this for both the one in the talking callback or the one in handle_cli_confbridge_kick or both (the fix is similar in both)? The one in handle_cli_confbridge_kick() applies to v11+. The other one applies to v12+. Richard -- _ -- 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
