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/7001-00000001?? [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=<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.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 -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
