[asterisk-users] SIP Calls Not Working

[Deepak Bhatia]

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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