"all Circuits are busy" On outgoing calls

Hi There,
My name is Joe (KiwifromNZ) I am from New Zealand this is my first post here.

I am a long term user of Asterisk (from the Trixbox days)
I have Asterisk Ver 11.4.0 on my machine. It has been working flawlessly for 2 years. I noticed a degrading of one line with an intermittent “all circuits are busy” on one line. It would fail over to the next line so it was not notices unless we were using more than 2 lines out.
I decided to upgrade and put the new version on. It is a complete clean install. The freepbx distro was dated 31 July 2013.
Now I have All incoming lines working (2 Voip Lines) but am unable to make any calls out greeted by the “all circuits are busy”

My provider does not support asterisk so will not look at any network details regarding my call attempts. In the log all I get is:

403 2013-08-05 09:24:59] WARNING[1894][C-000000d2] chan_sip.c: Received response: “Forbidden” from ‘sip:[email protected];tag=as23ae0fb5’

I have tried dial plan changes, Checked network routing which appears to be fine. I tested with X-Lite softphone on a machine, works fine, Incoming and outgoing. That generally says routing is ok?

I know how to get around simple practical faults i.e sound issues nat etc but this one is really got me.
I am finally stumped. Asterisk is running on an IBM Xserver 346 with Raid, 4 Gb Ram. I have Linksys SPA942 Phones connected and Linksys ATA analog boxs.
16 extensions, 2 trunks. Including an IAX phone accross the city connected in fine.
Now my log looks like hieroglyphics to me, I am hoping that it makes sense to someone here :slight_smile:
Because this is way above me I have basically guessed this is the problem area within the log by the “tag at the end of the forbidden line”

I apologize now if this message is not formatted correctly for the forum users its my first post.
I have checked the forum over the last 2 days for similar faults and dont seem to have found any. I thought it may be codec related but after changing codec settings the fault remains.

Regards
Joe

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK21f3d64f;rport
Max-Forwards: 70
From: sip:[email protected];tag=as30228e58
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Date: Sun, 04 Aug 2013 20:56:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1640260105 1640260105 IN IP4 192.168.1.5
s=Asterisk PBX 11.4.0
c=IN IP4 192.168.1.5
t=0 0
m=audio 18384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2013-08-05 08:56:28] VERBOSE[22661][C-000000d0] app_dial.c: – Called SIP/Christchurch/3382107
[2013-08-05 08:56:29] VERBOSE[1894] chan_sip.c:
<— SIP read from UDP:203.184.16.89:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.5:5060;received=222.XXX.XX.63;branch=z9hG4bK21f3d64f;rport=49154
From: sip:[email protected]:5060;tag=as30228e58
To: sip:[email protected]
Call-ID: [email protected]:5060
CSeq: 102 INVITE

<------------->
[2013-08-05 08:56:29] VERBOSE[1894] chan_sip.c: — (6 headers 0 lines) —
[2013-08-05 08:56:29] VERBOSE[1894] chan_sip.c:
<— SIP read from UDP:203.184.16.89:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.5:5060;received=222.XXX.XX.63;branch=z9hG4bK21f3d64f;rport=49154
From: sip:[email protected]:5060;tag=as30228e58
To: sip:[email protected];tag=aprqngfrt-ljaodv20000c6
Call-ID: [email protected]:5060
CSeq: 102 INVITE

<------------->
[2013-08-05 08:56:29] VERBOSE[1894] chan_sip.c: — (6 headers 0 lines) —
[2013-08-05 08:56:29] VERBOSE[1894][C-000000d0] chan_sip.c: Transmitting (NAT) to 203.184.16.89:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK21f3d64f;rport
Max-Forwards: 70
From: sip:[email protected];tag=as30228e58
To: sip:[email protected];tag=aprqngfrt-ljaodv20000c6
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.4.0)
Content-Length: 0


[2013-08-05 08:56:29] WARNING[1894][C-000000d0] chan_sip.c: Received response: “Forbidden” from ‘sip:[email protected];tag=as30228e58’

This was an interesting problem specific to my service provider in New Zealand.
They have appeared to have actively changed their proxy server to discourage Asterisk software users. Thus why the sanctioned X-lite and inksys boxes work.
Here is the change needed in the Sip Trunk:

Original:
disallow=all
username=643974XXXX
type=friend
secret=(your P/W)
qualify=yes
nat=yes
insecure=port,invite
host=chc.italk.co.nz
fromuser=643974XXXX
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw&alaw

New Details with substituted italk IP address for the “chc.italk.co.nz

disallow=all
username=643974XXXX
type=friend
secret=(your Pw)
qualify=yes
nat=yes
insecure=port,invite
host=203.184.16.2
fromuser=643974XXXX
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw&alaw

For my NZ colleagues…
No wonder I could not find much on it in the forum. I found details on this link (a local NZ forum)http://www.geekzone.co.nz/forums.asp?forumid=43&topicid=125724
On about page 3-4

Regards
Joe
KiwifromNZ