[asterisk-users] SIP Calls Not Working

[Hashmat Khan]

the warning message
"[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
Packet timed out after 32000ms with no response" 
generally suggest some network issues. if you do tcpdump / ethereal trace you will get a much better idea whats going on.
most probably you are not getting any response back to your INVITE , hence timerb kickin after 32sec and generate an autocongest



Date: Mon, 1 Sep 2014 19:30:21 +0530
From: deepak at voxomos.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] SIP Calls Not Working

== 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
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).
    -- 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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