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