I posted a query at trixbox and was asked to post log snippet over here. Here I am, I have no trouble routing DID for some test accounts I have but just can’t get Zingotel to route except to the main number. Here is a bit of log from one DID dialed, I didn’t do other number tests here yet as they always go to the same extension.
9059631819 is the main/account number.
It got CID no problem but thinks I dialed the main number instead of the virtual number.
I dialed 9059631832 which should have run my extension but instead it went to the inbound route for the main number that is the echo test.
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[Jan 16 22:15:59] VERBOSE[7263] logger.c: — (0 headers 0 lines) Nat keepalive —
[Jan 16 22:16:00] VERBOSE[7263] logger.c:
<— SIP read from 69.25.48.83:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 69.25.48.83;branch=z9hG4bKfdfe1c09fdfec896
From: “Deschamps P” sip:19053400406@flgw;tag=cba-8d3a-478ec896
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1200539799 INVITE
Contact: “Deschamps P” sip:[email protected]
Date: Thu, 17 Jan 2008 03:16:38 GMT
User-Agent: BRSIP v2.0.1.2
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
Allow-Events: keep-alive, message-summary
Supported: timer
Session-Expires: 600
Min-SE: 120
Expires: 300
Content-Type: application/sdp
Content-Length: 279
v=0
o=BRSDP 1672100 1672100 IN IP4 209.167.234.100
s=BRSDP Session
c=IN IP4 209.167.234.100
t=0 0
m=audio 7500 RTP/AVP 18 4 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[Jan 16 22:16:00] VERBOSE[7263] logger.c: — (18 headers 12 lines) —
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Sending to 69.25.48.83 : 5060 (no NAT)
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Using INVITE request as basis request - [email protected]
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found no matching peer or user for ‘69.25.48.83:5060’
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found RTP audio format 18
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found RTP audio format 4
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found RTP audio format 0
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found RTP audio format 8
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found RTP audio format 101
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Peer audio RTP is at port 209.167.234.100:7500
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found audio description format G729 for ID 18
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found audio description format G723 for ID 4
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found audio description format PCMU for ID 0
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found audio description format PCMA for ID 8
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Found audio description format telephone-event for ID 101
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (no
thing), combined - 0xc (ulaw|alaw)
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), c
ombined - 0x1 (telephone-event)
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Peer audio RTP is at port 209.167.234.100:7500
[Jan 16 22:16:00] VERBOSE[7263] logger.c: Looking for 9059631819 in from-sip-external (domain 204.101.242.93)
[Jan 16 22:16:00] VERBOSE[7263] logger.c: list_route: hop: sip:[email protected]
[Jan 16 22:16:00] VERBOSE[7263] logger.c:
<— Transmitting (no NAT) to 69.25.48.83:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.25.48.83;branch=z9hG4bKfdfe1c09fdfec896;received=69.25.48.83
From: “Deschamps P” sip:19053400406@flgw;tag=cba-8d3a-478ec896
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 1200539799 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0
<------------>
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-sip-external:1] NoOp(“SIP/flgw-08894388”, “Received inco
ming SIP connection from unknown peer to 9059631819”) in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-sip-external:2] Set(“SIP/flgw-08894388”, “DID=9059631819
”) in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-sip-external:3] Goto(“SIP/flgw-08894388”, “s|1”) in new
stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Goto (from-sip-external,s,1)
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/flgw-08894388”, “1?from-trunk|9059631
819|1”) in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Goto (from-trunk,9059631819,1)
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-trunk:1] Set(“SIP/flgw-08894388”, “__FROM_DID=9059631819
”) in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-trunk:2] GotoIf(“SIP/flgw-08894388”, “1 ?cidok”) in new
stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Goto (from-trunk,9059631819,4)
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-trunk:4] NoOp(“SIP/flgw-08894388”, “CallerID is “Descham
ps P” <19053400406>”) in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [9059631819@from-trunk:5] Goto(“SIP/flgw-08894388”, “ext-miscdests|1|1”)
in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Goto (ext-miscdests,1,1)
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [1@ext-miscdests:1] NoOp(“SIP/flgw-08894388”, “MiscDest: EchoTest”) in ne
w stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [1@ext-miscdests:2] Goto(“SIP/flgw-08894388”, “from-internal|*43|1”) in n
ew stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Goto (from-internal,*43,1)
[Jan 16 22:16:00] VERBOSE[7321] logger.c: – Executing [*43@from-internal:1] Answer(“SIP/flgw-08894388”, “”) in new stack
[Jan 16 22:16:00] VERBOSE[7321] logger.c: Audio is at 204.101.242.93 port 14410
[Jan 16 22:16:00] VERBOSE[7321] logger.c: Adding codec 0x4 (ulaw) to SDP
[Jan 16 22:16:00] VERBOSE[7321] logger.c: Adding codec 0x8 (alaw) to SDP
[Jan 16 22:16:00] VERBOSE[7321] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 16 22:16:00] VERBOSE[7321] logger.c: