[asterisk-users] SIP Calls Not Working

[Hashmat Khan]

what do you get on the asterisk console output ?

Date: Mon, 1 Sep 2014 18:53:51 +0530
From: deepak at voxomos.com
To: asterisk-users at 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)

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