Re: [asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example
Hi, You also can see example script for create cluster. https://github.com/netaskd/AFDINbeat --- Dmitry Burilov -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Thursday, September 19, 2013 10:34 PM To: Asterisk Users List Subject: Re: [asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example Be careful with DRDB singe failing drive/corruption on one peers takes down the other too... Check out haast as well (at www.generationd.com) for a commercial asterisk clustering solution. Michelle (GenerationD Systems) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko [asannu...@gmail.com] Sent: Thursday, September 19, 2013 10:24 AM To: Asterisk Users List Subject: Re: [asterisk-users] Looking for Asterisk+Pacemaker+Corosync+DRBD example Hello Edwards you can install fedora repositories and the HeartBeat from those repositories. If the failover is only for two servers, this is a good solution. In the directory list, you have to add /etc/dahdi (is you use dahdi) and /var/spool/asterisk Regards El 19/09/2013 08:58, Steve Edwards escribió: I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker, Corosync, and DRBD. All the examples I've found so far use Heartbeat, but Heartbeat is not in the repositories and doesn't want to compile from source. Does anyone have a working configuration they can share or a tutorial they can point me to? Also, what does drbdlinks bring to the party? Isn't just linking the 'top level' directories (/etc/asterisk/, /var/lib/asterisk/, /var/lib/mysql, etc) sufficient? -- _ -- 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] Asterisk high load when streaming MOH
Hi list, I've always about 50 concurrent SIP callers listening to several MOH streams (fed via mpg123) on Asterisk 11.4.0, 4x 2.2 GHz and the CPU usage is always at about 250% causing stuttering of the streams and delays when using Read() and Playback() (overall everything is just really slow/delayed). I've seen Asterisk doing that when the MOH stream that was fed via mpg123 was down... then it will just go nuts. But I'm pretty sure that all streams are active (checked via strace). My question is what would be the best way to look at the internals of Asterisk to really find out what exactly is causing that high CPU load? Thank you! Markus -- _ -- 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] Somewhat-OT: Stupid NAT tricks to learn from Apple?
I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As SDP bodies swell more and more can we hope to build significant support for multiplexing and deflate compression in the SIP-focused open source ecosystem? -- Kristian Kielhofner -- _ -- 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] Asterisk high load when streaming MOH
Sometimes I see something similar on my systems when several (3 or more) MOH streams are fed by mpg123. Suddenly some or all streams are stuttering, but the CPU load doesn't seem to go up significantly. Have you looked at what is happening with the receive queue of mpg123 (pgrep mpg123 and netstat -t -p)? I have written a small daemon for myself that watches the receiving tcp streams and if nothing seems to be happening, simply touches musiconhold.conf and issues an asterisk -rx \moh reload\. Not nice, but it works, eventually. When I listen to the same music streams I find that they are not as stable as one would wish, at least at a time scale of several hours. When using several moh file classes, I have never hat audio problems. jg -- _ -- 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] sipgate outgoing calls
Probably worth noting that sipgate will close (at least in the U.S.) on Oct. 31: http://www.besttechie.com/2013/09/13/voip-provider-sipgate-will-close-oct-31/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karsten Wemheuer Sent: Thursday, September 19, 2013 10:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sipgate outgoing calls Hi, Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb gpxctawjc...@irational.org: Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line What Sipgate product are You using? At least in Germany there are different configurations for the different products necessary. For Sipgate trunking and Sipgate team You have to configure an outboundproxy (which differs between both products). For Sipgate Basic you don't need an outboundproxy. As far as I remember there was an issue with some asterisk versions and the outboundproxy for Sipdate team. Karsten -- _ -- 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] Asterisk high load when streaming MOH
Am 20.09.2013 15:17, schrieb jg: Sometimes I see something similar on my systems when several (3 or more) MOH streams are fed by mpg123. Suddenly some or all streams are stuttering, but the CPU load doesn't seem to go up significantly. My CPU load is permanent at 200-250%. I have 7 active mpg123 streams. On another box running Asterisk 10.8.0, 4x 2.4 GHz, I have 49 active streams, but they get fed via mplayer instead of mpg123. There the load is close to zero, but I have to say that I have only 1-2 concurrent callers there. Strange! Have you looked at what is happening with the receive queue of mpg123 (pgrep mpg123 and netstat -t -p)? netstat -t -p shows me each of the 7 IP addresses and Recv-Q has some bytes listed (55000 - 113000) for each IP. According to man netstat Recv-Q means The count of bytes not copied by the user program connected to this socket. Note the not. Now, is that good or bad? I have written a small daemon for myself that watches the receiving tcp streams and if nothing seems to be happening, simply touches musiconhold.conf and issues an asterisk -rx \moh reload\. Not nice, but it works, eventually. When I listen to the same music streams I find that they are not as stable as one would wish, at least at a time scale of several hours. What I do is simply call killall mpg123 every 10 minutes via cron. Asterisk MOH will restart the process immediately and there will be less than a second of silence for the listener. Not that elegant but it helps with hanging audio streams... When using several moh file classes, I have never hat audio problems. Me neither. Thanks! Markus -- _ -- 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] The call is established but without exchanged voice packets
Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] sectioncontext=internalallowguest=noallowoverlap=notransport=udpbindport=5060bindaddr=0.0.0.0directmedia=nosrvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=IPlocalnet=172.16.0.255/255.255.255.0 The error messages [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+RetransmissionsPacket timed out after 32000ms with no response[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).[Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks. Date: Thu, 19 Sep 2013 13:14:59 +0500 From: msalman...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here!http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -- _ -- 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 -- Regards ** Muhammad Salman *** -- _ -- 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] Somewhat-OT: Stupid NAT tricks to learn from Apple?
Hi Kristian, On 09/20/2013 03:17 PM, Kristian Kielhofner wrote: I've been spending some time looking at some of the significant changes Apple has made to Facetime in iOS 7. I'm far from an Apple fanboy but some of them are pretty interesting: - multiplexing everything over a single UDP port - deflate compression with SIP - various /slight/ protocol violations ;) More here: http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html As SDP bodies swell more and more can we hope to build significant support for multiplexing and deflate compression in the SIP-focused open source ecosystem? Thanks for sharing your analysis. Interesting read. Makes me wonder why not more vendors/projects are doing port multiplexing. Let's hope it will pick up steam now that Apple has implemented it. Regards, Patrick -- _ -- 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] The call is established but without exchanged voice packets
Hello, If Asterisk version is 1.6 use nat=force_rport,comedia On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] section context=internal allowguest=no allowoverlap=no transport=udp bindport=5060 bindaddr=0.0.0.0 directmedia=no srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP localnet=172.16.0.255/255.255.255.0 The error messages [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks. -- Date: Thu, 19 Sep 2013 13:14:59 +0500 From: msalman...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine. The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here! http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -- _ -- 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 -- Regards ** Muhammad Salman *** -- _ -- 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:
Re: [asterisk-users] The call is established but without exchanged voice packets
Hello, I have Asterisk 1.8.10.1Moving to nat=force_rport,comedia hasn't solved the problem. Still having the same error! I am not sure if this is related to the problem here, but I was trying to test my voicemail and got this error No audio available).[Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)[Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No audio available on SIP/7001-0001??[Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Thanks. Date: Fri, 20 Sep 2013 16:05:35 +0200 From: asghar...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Hello,If Asterisk version is 1.6 use nat=force_rport,comedia On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] section context=internalallowguest=noallowoverlap=notransport=udpbindport=5060bindaddr=0.0.0.0directmedia=nosrvlookup=nodisallow=all allow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=IPlocalnet=172.16.0.255/255.255.255.0 The error messages [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks. Date: Thu, 19 Sep 2013 13:14:59 +0500 From: msalman...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here!http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -- _ -- 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 -- Regards ** Muhammad Salman *** -- _ -- 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
Re: [asterisk-users] Asterisk high load when streaming MOH
My CPU load is permanent at 200-250%. I have 7 active mpg123 streams. I forgot something. Even if your 7 streams are mp3 streams they cannot consume the CPU power you are seeing. -- _ -- 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] The call is established but without exchanged voice packets
Hello, i think your logic is wrong please explain me what are you trying to do? [internal] exten = 7002,1,Answer() exten = 7002,n,Playback(vm-nobodyavail) exten = 7002,n,Hangup() exten = 7001,1,Dial(SIP/7001,60) exten = 7001,n,Hangup() try this dial 7002 and you should listen vm-nobodyavail or 7001 to 7001 extension. On Fri, Sep 20, 2013 at 4:31 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, Here is my extension context, [internal] exten = 7001,1,Answer() exten = 7001,2,Dial(SIP/7001,60) exten = 7001,3,Playback(vm-nobodyavail) exten = 7001,4,VoiceMail(7001@main) ;forward to voicemail mailbox exten = 7001,5,Hangup() exten = 7002,1,Answer() exten = 7002,2,Dial(SIP/7002,60) exten = 7002,3,Playback(vm-nobodyavail) exten = 7002,4,VoiceMail(7002@main) exten = 7002,5,Hangup() exten = 7003,1,Answer() exten = 7003,2,Dial(SIP/7003,60) exten = 7003,3,Playback(vm-nobodyavail) exten = 7003,4,VoiceMail(7003@main) exten = 7003,5,Hangup() exten = 8001,1,VoicemailMain(7001@main) ;voicemail retreival exten = 8001,2,Hangup() exten = 8002,1,VoicemailMain(7002@main) exten = 8002,2,Hangup() -- Date: Fri, 20 Sep 2013 16:25:42 +0200 From: asghar...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Hello, paste you extension context. On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I have Asterisk 1.8.10.1 Moving to nat=force_rport,comedia hasn't solved the problem. Still having the same error! I am not sure if this is related to the problem here, but I was trying to test my voicemail and got this error No audio available). [Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No audio available on SIP/7001-0001?? [Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Thanks. -- Date: Fri, 20 Sep 2013 16:05:35 +0200 From: asghar...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Hello, If Asterisk version is 1.6 use nat=force_rport,comedia On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] section context=internal allowguest=no allowoverlap=no transport=udp bindport=5060 bindaddr=0.0.0.0 directmedia=no srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP localnet=172.16.0.255/255.255.255.0 The error messages [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks. -- Date: Thu, 19 Sep 2013 13:14:59 +0500 From: msalman...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine. The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this
[asterisk-users] Astricon - let's talk call centers?
Hi list, I know it's a bit OT, but for those who will be at the Astricon, we are organizing a very informal meeting (maybe in front of a pint or two) to talk about Asterisk for call-centers. No marketing or anything - just a way to exchange ideas and meet f2f. I created a facebook group to organize it - see https://www.facebook.com/groups/507826572618269/ See you in Atlanta! l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com -- _ -- 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] The call is established but without exchanged voice packets
Asmaa, You're getting ahead of yourself. How do you expect audio to work if your firewall/NAT settings aren't even configured correctly to establish SIP sessions? Go back and read the message that I sent yesterday. Fix the SIP three-way handshake problem. That is step 1 and you'll know you have it right when you stop seeing 'Retransmission timeout reached on transmission' errors. You still won't have audio but that's step 2. It requires properly configuring Asterisk's NAT settings and the firewall(s) between the phones and the server to allow RTP traffic to flow, but don't worry about it until step 1 is complete. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] The call is established but without exchanged voice packets
Hello, Here is my extension context, [internal]exten = 7001,1,Answer()exten = 7001,2,Dial(SIP/7001,60)exten = 7001,3,Playback(vm-nobodyavail)exten = 7001,4,VoiceMail(7001@main) ;forward to voicemail mailboxexten = 7001,5,Hangup() exten = 7002,1,Answer()exten = 7002,2,Dial(SIP/7002,60)exten = 7002,3,Playback(vm-nobodyavail)exten = 7002,4,VoiceMail(7002@main)exten = 7002,5,Hangup() exten = 7003,1,Answer()exten = 7003,2,Dial(SIP/7003,60)exten = 7003,3,Playback(vm-nobodyavail)exten = 7003,4,VoiceMail(7003@main)exten = 7003,5,Hangup() exten = 8001,1,VoicemailMain(7001@main) ;voicemail retreivalexten = 8001,2,Hangup() exten = 8002,1,VoicemailMain(7002@main)exten = 8002,2,Hangup() Date: Fri, 20 Sep 2013 16:25:42 +0200 From: asghar...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Hello,paste you extension context. On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have Asterisk 1.8.10.1Moving to nat=force_rport,comedia hasn't solved the problem. Still having the same error! I am not sure if this is related to the problem here, but I was trying to test my voicemail and got this error No audio available). [Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)[Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No audio available on SIP/7001-0001?? [Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Thanks. Date: Fri, 20 Sep 2013 16:05:35 +0200 From: asghar...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Hello,If Asterisk version is 1.6 use nat=force_rport,comedia On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] section context=internalallowguest=noallowoverlap=notransport=udpbindport=5060bindaddr=0.0.0.0directmedia=nosrvlookup=nodisallow=all allow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=IPlocalnet=172.16.0.255/255.255.255.0 The error messages [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response[Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks. Date: Thu, 19 Sep 2013 13:14:59 +0500 From: msalman...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is
Re: [asterisk-users] Asterisk high load when streaming MOH
My CPU load is permanent at 200-250%. I have 7 active mpg123 streams. My understanding is that the mpg123 stream gets decoded only a single time regardless of the number of listeners. netstat -t -p shows me each of the 7 IP addresses and Recv-Q has some bytes listed (55000 - 113000) for each IP. According to man netstat Recv-Q means The count of bytes not copied by the user program connected to this socket. Note the not. Now, is that good or bad? I see about the same values. The values should be varying and as low as possible. So 0 for a couple of minutes could mean that the stream is dead. What I do is simply call killall mpg123 every 10 minutes via cron. Asterisk MOH will restart the process immediately and there will be less than a second of silence for the listener. Not that elegant but it helps with hanging audio streams... My solution is essentially the same, only the other way around and based on whether a stream seems to be dead. moh reload essentially kills any mpg123 instances and restarts them. I was just too lazy to study the Asterisk sources carefully. When skimming the code there were quite a few Uh Oh. ... comments, and I assumed that moh reload is probably a safer thing to do than just killing mgp123. My system issues an moh reload maybe once or twice a day, and sometimes not even a single time. When I use Shoutcast streams there a a lot more restarts a day. jg -- _ -- 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] The call is established but without exchanged voice packets
Hello, paste you extension context. On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have Asterisk 1.8.10.1 Moving to nat=force_rport,comedia hasn't solved the problem. Still having the same error! I am not sure if this is related to the problem here, but I was trying to test my voicemail and got this error No audio available). [Sep 20 14:05:41] WARNING[11424]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Sep 20 14:05:54] WARNING[11424]: app.c:855 __ast_play_and_record: No audio available on SIP/7001-0001?? [Sep 20 14:06:13] WARNING[11387]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkNTY0YzZjMTcwNzcwYTg0NWRiMjlhYzQ4ZjFkOTc for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Thanks. -- Date: Fri, 20 Sep 2013 16:05:35 +0200 From: asghar...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Hello, If Asterisk version is 1.6 use nat=force_rport,comedia On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] section context=internal allowguest=no allowoverlap=no transport=udp bindport=5060 bindaddr=0.0.0.0 directmedia=no srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP localnet=172.16.0.255/255.255.255.0 The error messages [Sep 20 13:51:32] NOTICE[10979]: chan_sip.c:24728 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 7002 [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Sep 20 13:52:27] WARNING[10979]: chan_sip.c:3670 retrans_pkt: Hanging up call OGU1NzgyMmVmNjU1NTBlYmNkMWIwOGEzOWRjNGYxYWU - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ). [Sep 20 13:52:44] WARNING[10965]: asterisk.c:3190 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) Thanks. -- Date: Thu, 19 Sep 2013 13:14:59 +0500 From: msalman...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Choose suitable NAT settings from sip.conf turn direct media in sip.conf or per peer off On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote: Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine. The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see this chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse externip=IP [7001] type=friend host=dynamic secret=123 context=internal [7002] type=friend host=dynamic secret=456 context=internal A snoop capture for my call is uploaded in the following link. I wonder if there is any missing configuration or plugin need to be set here! http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 http://www.fileconvoy.com/dfl.php?id=gc0957418962ed157999374318118ae80b9d015992 Thanks. -- _ -- 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 -- Regards ** Muhammad Salman *** --
Re: [asterisk-users] The call is established but without exchanged voice packets
Hi Matthew, Indeed I missed your previous message!After changing the externip, it worked successfully... The sip session is established with the complete three-way handshake, and the voice packet is exchanged with no problem! Many thanks. Date: Fri, 20 Sep 2013 10:01:52 -0500 From: mr...@imminc.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] The call is established but without exchanged voice packets Asmaa, You're getting ahead of yourself. How do you expect audio to work if your firewall/NAT settings aren't even configured correctly to establish SIP sessions? Go back and read the message that I sent yesterday. Fix the SIP three-way handshake problem. That is step 1 and you'll know you have it right when you stop seeing 'Retransmission timeout reached on transmission' errors. You still won't have audio but that's step 2. It requires properly configuring Asterisk's NAT settings and the firewall(s) between the phones and the server to allow RTP traffic to flow, but don't worry about it until step 1 is complete. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] The call is established but without exchanged voice packets
Asmaa Ahmed wrote: Indeed I missed your previous message! After changing the externip, it worked successfully... The sip session is established with the complete three-way handshake, and the voice packet is exchanged with no problem! Many thanks. Asmaa, That's great news!! I guess the firewall settings were already correct and it was just a matter of configuring Asterisk properly. In my experience, the first call is always the hardest one to get working. Now that you've done that you can really start seeing what Asterisk can do. Have fun, but remember to take it step by step and don't hesitate to ask the list if you run into any problems. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Asterisk high load when streaming MOH
Am 20.09.2013 16:37, schrieb jg: My CPU load is permanent at 200-250%. I have 7 active mpg123 streams. I forgot something. Even if your 7 streams are mp3 streams they cannot consume the CPU power you are seeing. I thought so. So, I need to dig deeper... but how? Some command to show the different Asterisk internal sub-processes stats would be nice to begin with. Like what exactly is consuming that much CPU. -- _ -- 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] AstDB Partial Replication?
Any takers? astdb is based off of version 1 BerkeleyDB. Googling shows: http://www.voip-info.org/wiki/view/Asterisk+database It has a section on basic replication. Doug -- _ -- 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] AstDB Partial Replication?
- Original Message - Is anyone aware of a way to replicate parts of the AstDB to another Asterisk install? For example, to export all CF entries on a FreePBX based system to another system running FreePBX, I might do: asterisk -rx 'database show' | grep CF This gives me a list of data, which I can rsync to another host to reimport using 'database put'. BUT, the problem comes in when I want to sync CF entries to/from both Asterisk systems. I seem to be having race conditions where an entry is removed on system A, but before that removal can sync to system B, we've already imported that to system A again. Does this make sense? TLDR; How do I sync AstDB entries between two hosts, in both directions, while maintaining data integrity? Any takers? --Tim -- _ -- 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
