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Posted to user@openmeetings.apache.org by "Jibon L. Costa" <ji...@gmail.com> on 2015/10/18 23:27:34 UTC
SIP Transport Disconnecting Continuously !!
Hi,
I was testing openmeeting & asterisk on a ubuntu VPS. I followed
everything from tutorial but when I get access to any room Transport
Disconnecting Continuously. The output of asterisk log:
Packet timed out after 32000ms with no response
== Manager 'openmeetings' logged on from 127.0.0.1
== Manager 'openmeetings' logged off from 127.0.0.1
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [4004@rooms-red5sip:1]
GotoIf("SIP/red5sip_user-00000071", "0?ok:notavail") in new stack
-- Goto (rooms-red5sip,4004,3)
-- Executing [4004@rooms-red5sip:3]
Hangup("SIP/red5sip_user-00000071", "") in new stack
== Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
'SIP/red5sip_user-00000071'
== Manager 'openmeetings' logged on from 127.0.0.1
== Manager 'openmeetings' logged off from 127.0.0.1
== Manager 'openmeetings' logged on from 127.0.0.1
== Manager 'openmeetings' logged off from 127.0.0.1
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [4004@rooms-red5sip:1]
GotoIf("SIP/red5sip_user-00000072", "0?ok:notavail") in new stack
-- Goto (rooms-red5sip,4004,3)
-- Executing [4004@rooms-red5sip:3]
Hangup("SIP/red5sip_user-00000072", "") in new stack
== Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
'SIP/red5sip_user-00000072'
== Manager 'openmeetings' logged on from 127.0.0.1
== Manager 'openmeetings' logged off from 127.0.0.1
[Oct 18 17:26:26] WARNING[1364]: chan_sip.c:4024 retrans_pkt:
Retransmission timeout reached on transmission
912799716351@167.114.221.90 for seqno 2 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
== Manager 'openmeetings' logged on from 127.0.0.1
== Manager 'openmeetings' logged off from 127.0.0.1
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [4004@rooms-red5sip:1]
GotoIf("SIP/red5sip_user-00000073", "0?ok:notavail") in new stack
-- Goto (rooms-red5sip,4004,3)
-- Executing [4004@rooms-red5sip:3]
Hangup("SIP/red5sip_user-00000073", "") in new stack
== Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
'SIP/red5sip_user-00000073'
== Manager 'openmeetings' logged on from 127.0.0.1
== Manager 'openmeetings' logged off from 127.0.0.1
[Oct 18 17:26:31] WARNING[1364]: chan_sip.c:4024 retrans_pkt:
Retransmission timeout reached on transmission
624926910521@167.114.221.90 for seqno 2 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
Any idea where did I do wrong?
--
Thanks & Regards
Jibon Lawrence Costa
Re: SIP Transport Disconnecting Continuously !!
Posted by "Jibon L. Costa" <ji...@gmail.com>.
Thanks but why this error:
.49.128
[Oct 19 14:38:39] WARNING[20592][C-0000001b]: res_odbc.c:641
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22:
[MySQL][ODBC 5.1
Driver][mysqld-5.5.44-MariaDB-1ubuntu0.14.04.1]Unknown column
'insecure' in 'where clause' (107)
[Oct 19 14:38:39] WARNING[20592][C-0000001b]: res_odbc.c:653
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
connection to asterisk [asterisk-connector]...
[Oct 19 14:38:39] WARNING[20592][C-0000001b]: res_odbc.c:757
ast_odbc_sanity_check: Connection is down attempting to reconnect...
[Oct 19 14:38:44] NOTICE[20592][C-0000001b]: res_odbc.c:1529
odbc_obj_connect: Connecting asterisk
[Oct 19 14:38:44] NOTICE[20592][C-0000001b]: res_odbc.c:1568
odbc_obj_connect: res_odbc: Connected to asterisk [asterisk-connector]
[Oct 19 14:38:44] WARNING[20592][C-0000001b]: res_odbc.c:641
ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42S22:
[MySQL][ODBC 5.1
Driver][mysqld-5.5.44-MariaDB-1ubuntu0.14.04.1]Unknown column
'insecure' in 'where clause' (107)
[Oct 19 14:38:44] WARNING[20592][C-0000001b]: res_odbc.c:653
ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying
connection to asterisk [asterisk-connector]...
[Oct 19 14:38:44] WARNING[20592][C-0000001b]: res_odbc.c:757
ast_odbc_sanity_check: Connection is down attempting to reconnect...
Specially: Unknown column 'insecure' in 'where clause' (107)?
On 10/20/15, Maxim Solodovnik <so...@gmail.com> wrote:
> I see no issues in this log :(
> Maybe others can help
>
> On Mon, Oct 19, 2015 at 3:27 AM, Jibon L. Costa <ji...@gmail.com>
> wrote:
>
>> Hi,
>>
>> I was testing openmeeting & asterisk on a ubuntu VPS. I followed
>> everything from tutorial but when I get access to any room Transport
>> Disconnecting Continuously. The output of asterisk log:
>>
>> Packet timed out after 32000ms with no response
>> == Manager 'openmeetings' logged on from 127.0.0.1
>> == Manager 'openmeetings' logged off from 127.0.0.1
>> == Using SIP VIDEO CoS mark 6
>> == Using SIP RTP CoS mark 5
>> -- Executing [4004@rooms-red5sip:1]
>> GotoIf("SIP/red5sip_user-00000071", "0?ok:notavail") in new stack
>> -- Goto (rooms-red5sip,4004,3)
>> -- Executing [4004@rooms-red5sip:3]
>> Hangup("SIP/red5sip_user-00000071", "") in new stack
>> == Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
>> 'SIP/red5sip_user-00000071'
>> == Manager 'openmeetings' logged on from 127.0.0.1
>> == Manager 'openmeetings' logged off from 127.0.0.1
>> == Manager 'openmeetings' logged on from 127.0.0.1
>> == Manager 'openmeetings' logged off from 127.0.0.1
>> == Using SIP VIDEO CoS mark 6
>> == Using SIP RTP CoS mark 5
>> -- Executing [4004@rooms-red5sip:1]
>> GotoIf("SIP/red5sip_user-00000072", "0?ok:notavail") in new stack
>> -- Goto (rooms-red5sip,4004,3)
>> -- Executing [4004@rooms-red5sip:3]
>> Hangup("SIP/red5sip_user-00000072", "") in new stack
>> == Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
>> 'SIP/red5sip_user-00000072'
>> == Manager 'openmeetings' logged on from 127.0.0.1
>> == Manager 'openmeetings' logged off from 127.0.0.1
>> [Oct 18 17:26:26] WARNING[1364]: chan_sip.c:4024 retrans_pkt:
>> Retransmission timeout reached on transmission
>> 912799716351@167.114.221.90 for seqno 2 (Critical Response) -- See
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>> Packet timed out after 31999ms with no response
>> == Manager 'openmeetings' logged on from 127.0.0.1
>> == Manager 'openmeetings' logged off from 127.0.0.1
>> == Using SIP VIDEO CoS mark 6
>> == Using SIP RTP CoS mark 5
>> -- Executing [4004@rooms-red5sip:1]
>> GotoIf("SIP/red5sip_user-00000073", "0?ok:notavail") in new stack
>> -- Goto (rooms-red5sip,4004,3)
>> -- Executing [4004@rooms-red5sip:3]
>> Hangup("SIP/red5sip_user-00000073", "") in new stack
>> == Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
>> 'SIP/red5sip_user-00000073'
>> == Manager 'openmeetings' logged on from 127.0.0.1
>> == Manager 'openmeetings' logged off from 127.0.0.1
>> [Oct 18 17:26:31] WARNING[1364]: chan_sip.c:4024 retrans_pkt:
>> Retransmission timeout reached on transmission
>> 624926910521@167.114.221.90 for seqno 2 (Critical Response) -- See
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>> Packet timed out after 31999ms with no response
>>
>> Any idea where did I do wrong?
>> --
>> Thanks & Regards
>> Jibon Lawrence Costa
>>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
Thanks & Regards
Jibon Lawrence Costa
Re: SIP Transport Disconnecting Continuously !!
Posted by Maxim Solodovnik <so...@gmail.com>.
I see no issues in this log :(
Maybe others can help
On Mon, Oct 19, 2015 at 3:27 AM, Jibon L. Costa <ji...@gmail.com>
wrote:
> Hi,
>
> I was testing openmeeting & asterisk on a ubuntu VPS. I followed
> everything from tutorial but when I get access to any room Transport
> Disconnecting Continuously. The output of asterisk log:
>
> Packet timed out after 32000ms with no response
> == Manager 'openmeetings' logged on from 127.0.0.1
> == Manager 'openmeetings' logged off from 127.0.0.1
> == Using SIP VIDEO CoS mark 6
> == Using SIP RTP CoS mark 5
> -- Executing [4004@rooms-red5sip:1]
> GotoIf("SIP/red5sip_user-00000071", "0?ok:notavail") in new stack
> -- Goto (rooms-red5sip,4004,3)
> -- Executing [4004@rooms-red5sip:3]
> Hangup("SIP/red5sip_user-00000071", "") in new stack
> == Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
> 'SIP/red5sip_user-00000071'
> == Manager 'openmeetings' logged on from 127.0.0.1
> == Manager 'openmeetings' logged off from 127.0.0.1
> == Manager 'openmeetings' logged on from 127.0.0.1
> == Manager 'openmeetings' logged off from 127.0.0.1
> == Using SIP VIDEO CoS mark 6
> == Using SIP RTP CoS mark 5
> -- Executing [4004@rooms-red5sip:1]
> GotoIf("SIP/red5sip_user-00000072", "0?ok:notavail") in new stack
> -- Goto (rooms-red5sip,4004,3)
> -- Executing [4004@rooms-red5sip:3]
> Hangup("SIP/red5sip_user-00000072", "") in new stack
> == Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
> 'SIP/red5sip_user-00000072'
> == Manager 'openmeetings' logged on from 127.0.0.1
> == Manager 'openmeetings' logged off from 127.0.0.1
> [Oct 18 17:26:26] WARNING[1364]: chan_sip.c:4024 retrans_pkt:
> Retransmission timeout reached on transmission
> 912799716351@167.114.221.90 for seqno 2 (Critical Response) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 31999ms with no response
> == Manager 'openmeetings' logged on from 127.0.0.1
> == Manager 'openmeetings' logged off from 127.0.0.1
> == Using SIP VIDEO CoS mark 6
> == Using SIP RTP CoS mark 5
> -- Executing [4004@rooms-red5sip:1]
> GotoIf("SIP/red5sip_user-00000073", "0?ok:notavail") in new stack
> -- Goto (rooms-red5sip,4004,3)
> -- Executing [4004@rooms-red5sip:3]
> Hangup("SIP/red5sip_user-00000073", "") in new stack
> == Spawn extension (rooms-red5sip, 4004, 3) exited non-zero on
> 'SIP/red5sip_user-00000073'
> == Manager 'openmeetings' logged on from 127.0.0.1
> == Manager 'openmeetings' logged off from 127.0.0.1
> [Oct 18 17:26:31] WARNING[1364]: chan_sip.c:4024 retrans_pkt:
> Retransmission timeout reached on transmission
> 624926910521@167.114.221.90 for seqno 2 (Critical Response) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 31999ms with no response
>
> Any idea where did I do wrong?
> --
> Thanks & Regards
> Jibon Lawrence Costa
>
--
WBR
Maxim aka solomax