Call disconnects after 4 rings - Please help

Hi All,

I appear to have my freepbx working fine apart from one issue.

If anyone can assist me, i would really appreciate your time.

Incoming calls ring 4 times then disconnect. If i answer within 4 rings, everything appears to work fine. I can transfer calls between phones indoors.

Below is some logs to view my basic set up and the pastebin log shows the call disconnect.

http://pastebin.com/KMXmrafe

And a couple of screenshots in my dropbox folder.

Asterisk (Ver. 11.6.0)
Linux raspbx 3.6.11+ #520 PREEMPT Wed Aug 7 16:07:34 BST 2013 armv6l

Thanks in advance

Mitch

To add some additional information. If i ring from one extension to another, the call does not disconnect. It only happens when calling from an external phone. Also, if i remove freepbx and have my yealink t20 phone just connected to a router, it will ring at least 30 times before disconnecting.

lastly, it also makes no difference if the phones are in a ring group or not.

I have copied the config in the url below.
I changed my nat settings to nat=yes and it gave me the error message below

[2014-03-09 13:53:05] WARNING[3010]: sip/config_parser.c:812 sip_parse_nat_option: nat=yes is deprecated, use nat=force_rport,comedia instead

So i changed it to nat=force_rport,comedia

I am currently using nat=no although i am behind a nat router.

It does not matter what config i seem to use, the incoming calls always disconnect on the 4th ring.

Has anyone come across this before ? Is it a common fault ? Is anyone able to point me in the right direction please.

Below is the code that i receive when the call disconnects.

– Goto (macro-dial-one,s,43)
– Executing [[email protected]:43] Dial(“SIP/localphone-inbound-00000026”, “SIP/401,tr”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/401
– SIP/401-00000027 is ringing
– SIP/401-00000027 is ringing
== Spawn extension (macro-dial-one, s, 43) exited non-zero on ‘SIP/localphone-inbound-00000026’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 7) exited non-zero on ‘SIP/localphone-inbound-00000026’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 401, 2) exited non-zero on ‘SIP/localphone-inbound-00000026’
– Executing [[email protected]:1] Macro(“SIP/localphone-inbound-00000026”, “hangupcall,”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/localphone-inbound-00000026”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [[email protected]:3] ExecIf(“SIP/localphone-inbound-00000026”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [[email protected]:4] Hangup(“SIP/localphone-inbound-00000026”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/localphone-inbound-00000026’ in macro ‘hangupcall’
== Spawn extension (from-did-direct, h, 1) exited non-zero on ‘SIP/localphone-inbound-00000026’
[2014-03-09 14:32:05] NOTICE[3010]: chan_sip.c:29376 sip_poke_noanswer: Peer ‘401’ is now UNREACHABLE! Last qualify: 123
raspbx*CLI>

I forgot to add the url

http://pbxinaflash.com/community/index.php?threads/localphone-configuration-in-piaf.7348/

ok, i have been unable to resolve this.
However, i have changed my trunk provider from localphone.com to sipgate.co.uk and the incoming calls ring normally.

I will leave my set-up using the following. Incoming calls to sipgate & outgoing using localphone.
it’s working, and for now that is good enough for me.