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] 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] 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
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] 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
[asterisk-users] The call is established but without exchanged voice packets
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 thischan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=IP[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=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
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
Re: [asterisk-users] The call is established but without exchanged voice packets
Asmaa Ahmed wrote: 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, a nd 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) The SIP trace you provided breaks down as follows: X-Lite Asterisk --- --- INVITE(No Auth) --- --- 401 Unauthorized ACK --- INVITE(Auth)--- --- 100 Trying --- 200 OK --- 200 OK (Retransmitted 10 Times) --- BYE OK --- This shows that the three-way handshake (INVITE/200 OK/ACK) used to establish SIP sessions is not completed because Asterisk never receives an ACK from X-Lite. After retransmitting the 200 OK 10 times Asterisk gives up and disconnects the call. 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 From the SIP trace, I believe 'externip=41.46.164.96' is set. If that is the case, try changing it to 'externip=54.241.129.14'. You should also set localnet as follows: ; RFC 1918 addresses localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 If that doesn't work you can also try setting 'nat=force_rport' instead of 'nat=yes'. [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 At this point, you should be able to establish a call between the two X-Lite phones that won't get disconnected due to failing to complete the three-way handshake. There may still not be voice because the firewall(s) between Asterisk and the X-Lite phones may block the RTP traffic. The phones appear to be on the same network, so you can try setting 'canreinvite=yes' to workaround this problem until the firewall(s) are configured to allow RTP traffic on the UDP port range specified in 'rtp.conf' (the default range is 1-2). Good luck and please report your progress back to the list. 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
