Hello, I have 2 customers who are authenticated by IP Address, and I'm sending for each one a DID number, both customers with the lastest Freepbx version, the thing is, i did setup the trunk in Freepbx like this in the PEER Details
Code:
type=peer
sendrpid=yes
qualify=yes
nat=yes
insecure=port,invite
host=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=gsm
After that, i just add the incoming DID number, but no luck, i just get this
Code:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [s@from-trunk:1] NoOp("SIP/Telefonia-00000011", "No DID or CID Match") in new stack
-- Executing [s@from-trunk:2] Answer("SIP/Telefonia-00000011", "") in new stack
-- Executing [s@from-trunk:3] Wait("SIP/Telefonia-00000011", "2") in new stack
-- Executing [s@from-trunk:4] Playback("SIP/Telefonia-00000011", "ss-noservice") in new stack
-- <SIP/Telefonia-00000011> Playing 'ss-noservice.gsm' (language 'en')
== Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/Telefonia-00000011'
-- Executing [h@from-trunk:1] Macro("SIP/Telefonia-00000011", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/Telefonia-00000011", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/Telefonia-00000011", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/Telefonia-00000011", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/Telefonia-00000011' in macro 'hangupcall'
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/Telefonia-00000011'
After some search on google, i just figure out that a2billing was not sending the DID number in the sip header, i did a debug on the peer and i saw this:
Code:
pruebas*CLI> sip set debug peer Telefonia
SIP Debugging Enabled for IP: 19x.2xx.1xx.2xx
<--- SIP read from UDP:19x.2xx.1xx.2xx:5060 --->
INVITE sip:16x.2xx.1xx.1xx SIP/2.0
Via: SIP/2.0/UDP 19x.2xx.1xx.2xx:5060;branch=z9hG4bK6480fe2d;rport
Max-Forwards: 70
From: "529981677544" <sip:
[email protected]>;tag=as4c594b09
To: <sip:1xx.2xx.1xx.1xx>
Contact: <sip:
[email protected]:5060>
Call-ID:
[email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.7.0)
Date: Fri, 03 Oct 2014 20:00:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 333
As you can see here: >
To: <sip:1xx.2xx.1xx.1xx>, that suppossed to be
To: <sip:[email protected]:48798;rinstance=0d570aa4a6d85d31;transport=UDPThis is not happening with a normal sip cliente, using registration, i hope you can help me, Thanks!