== Using SIP RTP CoS mark 5 -- Executing [100@exten-101:1] Dial("SIP/101-00000014", "SIP/100") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/100 -- Registered SIP '101' at 115.252.66.70:55258 [Sep 1 18:10:20] NOTICE[4629]: chan_sip.c:25735 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101 [Sep 1 18:10:43] WARNING[4629]: chan_sip.c:3982 retrans_pkt: Retransmission timeout reached on transmission 5f0235b842799d285a70eb2d452974fb@dynamic for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions <https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions> Packet timed out after 32000ms with no response [Sep 1 18:10:43] WARNING[4629]: chan_sip.c:4011 retrans_pkt: Hanging up call 5f0235b842799d285a70eb2d452974fb@dynamic - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions <https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>). -- SIP/100-00000015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/101-00000014' status is 'CONGESTION'
Regards Deepak Bhatia Software Consultant Voxomos Systems Pvt. Limited Mobile: 91 9811196957 C56A/27, Sector 62, NOIDA (NCR), UP, India Skype: toreachdeepak On Mon, Sep 1, 2014 at 7:26 PM, Hashmat Khan <hyk...@hotmail.com> wrote: > what do you get on the asterisk console output ? > > ------------------------------ > Date: Mon, 1 Sep 2014 18:53:51 +0530 > From: dee...@voxomos.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] SIP Calls Not Working > > > Hello, > > I have two sip phones (zoiper). Earlier these used to communicate using > the settings below for sip.conf and extensions.conf and now we asterisk > 1.8.29.0, so these phones have stopped communicating. My question is that > does 1.8.29.0 release require any more changes to be done to the sip.conf > and extensions.conf to make the below work ? > > The sip.conf contains following enteries > ================================== > [100] > type=friend > username=100 > secret=100 > host=dynamic > port=5060 > dtmfmode=rfc2833 > fromdomain=dynamic > nat=no > canreinvite=false > context=exten-100 > > [101] > type=friend > username=101 > secret=101 > host=dynamic > port=5060 > dtmfmode=rfc2833 > fromdomain=dynamic > nat=no > canreinvite=false > context=exten-101 > > The extensions.conf contains > ======================== > > [exten-100] > exten => 101,1,Dial(SIP/101) > ;exten => echo,1,Echo() > ;exten => busytone,1,Playback(moh) > ;exten => 101,n,Hangup() > exten => 100,1,Answer() > exten => 100,n,Hangup() > > [exten-101] > exten => 101,1,Answer() > exten => 101,n,Hangup() > exten => 100,1,Dial(SIP/100) > ;exten => _x.,1,Playback(moh) > > -- _____________________________________________________________________ > -- 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