Topic: Extensions cut off after a few minutes

[del]

Hi All,

Just recently when I answer an incoming call on any extension it cuts off after a minute or so. Any help is appreciated. I am running release # 3.1.0-128. I have 1 softphone extension and a PAP2 with a Seimens Gigaset handset. Thanks in advance.

Del

[SARK devs]

We'll need a bit more than that to help you Del.  A good start would be a view of the call on the Asterisk console.  If it was running OK and now it isn't then you need to consider what has changed in the interim. 

Kind Regards

S

[del]

This is the Asterisk Console output when I called in to the PBX on my mobile:

[root@sark-pbx ~]# asterisk -rvvvvv
Asterisk 1.8.5.0, Copyright (C) 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail                                                                                                                                                             s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.8.5.0 currently running on sark-pbx (pid = 32047)
Verbosity was 0 and is now 5
  == Using SIP RTP CoS mark 5
    -- Executing [2585845@mainmenu:1] AGI("SIP/peer568-00000055", "sarkhpe,Inbound,2585845") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=2585845)
    -- <SIP/peer568-00000055>AGI Script sarkhpe completed, returning 0
    -- Executing [501@extensions:1] AGI("SIP/peer568-00000055", "sarkhpe,Alias,SIP/402 SIP/401,501") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Dial) Options: (SIP/402&SIP/401,15,)
  == Using SIP RTP CoS mark 5
    -- Called SIP/402
  == Using SIP RTP CoS mark 5
    -- Called SIP/401
    -- SIP/402-00000056 connected line has changed. Saving it until answer for SIP/peer568-00000055
    -- SIP/401-00000057 connected line has changed. Saving it until answer for SIP/peer568-00000055
    -- SIP/402-00000056 is ringing
    -- SIP/401-00000057 is ringing
    -- SIP/402-00000056 connected line has changed. Saving it until answer for SIP/peer568-00000055
    -- SIP/402-00000056 answered SIP/peer568-00000055
    -- Locally bridging SIP/peer568-00000055 and SIP/402-00000056
[Nov 18 15:05:23] WARNING[32071]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 78988fba7666076d65a48f192854dc69@sipgate.co.uk for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Nov 18 15:05:23] WARNING[32071]: chan_sip.c:3651 retrans_pkt: Hanging up call 78988fba7666076d65a48f192854dc69@sipgate.co.uk - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Executing [h@extensions:1] Hangup("SIP/peer568-00000055", "") in new stack
  == Spawn extension (extensions, h, 1) exited non-zero on 'SIP/peer568-00000055'
    -- <SIP/peer568-00000055>AGI Script sarkhpe completed, returning 4
  == Spawn extension (extensions, 501, 1) exited non-zero on 'SIP/peer568-00000055'
       > doing dnsmgr_lookup for 'sipgw.voicenetwork.ca'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgw.voicenetwork.ca' mapped to host sipgw.voicenetwork.ca, port 5060
       > doing dnsmgr_lookup for 'did.voicenetwork.ca'
       > doing dnsmgr_lookup for 'sipgw.voicenetwork.ca'
       > ast_get_srv: SRV lookup for '_sip._udp.sipgw.voicenetwork.ca' mapped to host sipgw.voicenetwork.ca, port 5060
       > doing dnsmgr_lookup for 'did.voicenetwork.ca'
sark-pbx*CLI>

I can see that it timed out but I don't really know why.

Cheers,
Del

[SARK devs]

This is usually a NAT problem

If you look at the link printed in your console output...

Code: [Select]

https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
...it will give you a much better explanation than I can.  Usually, this is caused by something you aren't doing correctly; either in Asterisk or in your firewall.  If you are running SAIL server-only then check that you have included the correct external IP address in globals.   Also check that your firewall is open to 5060 and 10000-20000 and that they are forwarded to SAIL.

Kind Regards

S

[del]

This is usually a NAT problem

If you look at the link printed in your console output...

Code: [Select]

https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
...it will give you a much better explanation than I can.  Usually, this is caused by something you aren't doing correctly; either in Asterisk or in your firewall.  If you are running SAIL server-only then check that you have included the correct external IP address in globals.   Also check that your firewall is open to 5060 and 10000-20000 and that they are forwarded to SAIL.

Kind Regards

S

Hi Jeff,

It was the external IP address, Virgin had changed it. Can I put a URL in there instead? That way dyndns.org will update it for me. thanks again for your help.

Regards,
Del

[SARK devs]

no, you must put an ip address there

Your Virgin dynamic IP should only change very very infrequently or if you change your router.

Kind Regards

S

[del]

Your Virgin dynamic IP should only change very very infrequently or if you change your router.

Kind Regards

S

This is the first time it's changed since February  Probably why it caught me out 

Del