Re: [asterisk-users] SIP Calls Not Working
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: dee...@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/101-0014", "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/100-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/101-0014' 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 wrote:
what do you get on the asterisk console output ?
Date: Mon, 1 Sep 2014 18:53:51 +0530
From: dee...@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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Re: [asterisk-users] SIP Calls Not Working
== Using SIP RTP CoS mark 5
-- Executing [100@exten-101:1] Dial("SIP/101-0014", "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
<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@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/100-0015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/101-0014' 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 wrote:
> what do you get on the asterisk console output ?
>
> --
> Date: Mon, 1 Sep 2014 18:53:51 +0530
> From: dee...@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)
>
> -- _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
> or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] SIP Calls Not Working
what do you get on the asterisk console output ? Date: Mon, 1 Sep 2014 18:53:51 +0530 From: dee...@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) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
