I see no issues in this log :(
Maybe others can help
On Mon, Oct 19, 2015 at 3:27 AM, Jibon L. Costa <[email protected]>
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
> [email protected] 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
> [email protected] 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
