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<body class='hmmessage'><div dir='ltr'>what do you get on the asterisk console output ?<br><br><div><hr id="stopSpelling">Date: Mon, 1 Sep 2014 18:53:51 +0530<br>From: deepak@voxomos.com<br>To: asterisk-users@lists.digium.com<br>Subject: [asterisk-users] SIP Calls Not Working<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>
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