[asterisk-users] SIP Calls Not Working

[Deepak Bhatia]

== Using SIP RTP CoS mark 5
-- Executing [100 at 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 at 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 at 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 <hykhan at hotmail.com> wrote:

> 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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