<div dir="ltr">== Using SIP RTP CoS mark 5<br> -- Executing [100@exten-101:1] Dial("SIP/101-00000014", "SIP/100") in new stack<br> == Using SIP RTP CoS mark 5<br> -- Called SIP/100<br> -- Registered SIP '101' at <a href="http://115.252.66.70:55258">115.252.66.70:55258</a><br>
[Sep 1 18:10:20] NOTICE[4629]: chan_sip.c:25735 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101<br>[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 <a class="" href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/ ... nsmissions</a><br>Packet timed out after 32000ms with no response<br>[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 <a class="" href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/ ... nsmissions</a>).<br> -- SIP/100-00000015 is circuit-busy<br> == Everyone is busy/congested at this time (1:0/1/0)<br>
-- Auto fallthrough, channel 'SIP/101-00000014' status is 'CONGESTION'</div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><div><div>Regards<br></div><div><br>Deepak Bhatia<br>Software Consultant<br>
Voxomos Systems Pvt. Limited<br></div>Mobile: 91 9811196957<br><span>C56A/27, Sector 62, NOIDA (NCR), UP, India</span><br></div><div>Skype: toreachdeepak<br></div></div></div>
<br><br><div class="gmail_quote">On Mon, Sep 1, 2014 at 7:26 PM, Hashmat Khan <span dir="ltr"><<a href="mailto:hykhan@hotmail.com" target="_blank">hykhan@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">what do you get on the asterisk console output ?<br><br><div><hr>Date: Mon, 1 Sep 2014 18:53:51 +0530<br>From: <a href="mailto:deepak@voxomos.com" target="_blank">deepak@voxomos.com</a><br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
Subject: [asterisk-users] SIP Calls Not Working<div><div class="h5"><br><br><div dir="ltr">Hello,<br><br>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 ?<br><br>The sip.conf contains following enteries<br>==================================<br>[100]<br>type=friend<br>username=100<br>secret=100<br>host=dynamic<br>port=5060<br>dtmfmode=rfc2833<br>fromdomain=dynamic<br>
nat=no<br>canreinvite=false<br>context=exten-100<br><br>[101]<br>type=friend<br>username=101<br>secret=101<br>host=dynamic<br>port=5060<br>dtmfmode=rfc2833<br>fromdomain=dynamic<br>nat=no<br>canreinvite=false<br>context=exten-101<br>
<br>The extensions.conf contains<br>========================<br><br>[exten-100]<br>exten => 101,1,Dial(SIP/101)<br>;exten => echo,1,Echo()<br>;exten => busytone,1,Playback(moh)<br>;exten => 101,n,Hangup()<br>
exten => 100,1,Answer()<br>
exten => 100,n,Hangup()<br><br>[exten-101]<br>exten => 101,1,Answer()<br>exten => 101,n,Hangup()<br>exten => 100,1,Dial(SIP/100)<br>;exten => _x.,1,Playback(moh)</div>
<br></div></div>--
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