SIP Trunking, trunkinbound and s exten
Posted: Tue Sep 15, 2009 3:41 pmI am trying to use my DID's with my SIP account. The account registers correctly as seen by 'sip show registry', and I can see in sip debug that the call is coming through.
The problem is that asterisk is looking for the 's' exten, and fails the call when it can't find it. The 'trunkinbound' context is from the vicidial default files.
I have an IAX trunk w/DID that works just fine, and routes through agi-DID_route.agi with no problem.
I know there is something that I need to change, but i don't know what it is. What makes sip go to s instead of using the inbound DID?
Here is the sip debug info:
extenions.conf:
sip-vicidial.conf:
The problem is that asterisk is looking for the 's' exten, and fails the call when it can't find it. The 'trunkinbound' context is from the vicidial default files.
I have an IAX trunk w/DID that works just fine, and routes through agi-DID_route.agi with no problem.
I know there is something that I need to change, but i don't know what it is. What makes sip go to s instead of using the inbound DID?
Here is the sip debug info:
Using INVITE request as basis request - e317ce94fd6628a4c48ec99a9dacc7b3-4aafd9e1@173.12.100.11
Sending to 65.59.218.99 : 5060 (NAT)
Found peer 'proximiti75'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 65.59.218.166:37758
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Looking for s in trunkinbound (domain 173.XXX.XXX.XXX)
Reliably Transmitting (NAT) to 65.59.218.99:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 65.59.218.99:5060;branch=z9hG4bK2092f9d93bc03eb307f8ccf68bade827-0;received=65.59.218.99
From: "FTLAUDER"<sip:19545795515@65.59.218.99:5060>;tag=b2232a3d366395c26fb22c74a1268c4c
To: <sip:19543629975@173.12.100.11:5060>;tag=as413af22f
Call-ID: e317ce94fd6628a4c48ec99a9dacc7b3-4aafd9e1@173.12.100.11
CSeq: 38918 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
extenions.conf:
[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)
; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
sip-vicidial.conf:
register=>19543629975:12345678@ast.proximiti.com
; VICIDIAL Carrier:
[954362XXXX]
context=trunkinbound
type=friend
host=ast.proximiti.com
username=1954362XXXX
fromuser=1954362XXXX
secret=12345678
trustrpid=yes
sendrpid=yes
nat=yes
canreinvite=no
insecure=port,invite
disallow=all
allow=ulaw
dtmfmode=inband