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[deepak at voxomos.com]

== Using SIP RTP CoS mark 5
-- Executing [100@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@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@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@hotmail.com (hykhan@hotmail.com)> wrote:

[hykhan at hotmail.com]

the warning message

"[Sep 1 18:10:43] WARNING[4629]: chan_sip.c:3982 retrans_pkt: Retransmission timeout reached on transmission 5f0235b842799d285a70eb2d452974fb@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@voxomos.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIP Calls Not Working

== Using SIP RTP CoS mark 5
-- Executing [100@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@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@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@hotmail.com (hykhan@hotmail.com)> wrote:

[hykhan at hotmail.com]

what do you get on the asterisk console output ?

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

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)