fzr600dave
asked on
Problem with accepting a SIP trunk on IP 500
we have just setup a sip trunk with a friendly provider but despite the IPO receiving the SIP INVITE it doesn't accept the call, the logs suggest URI mapping but I cannot see whats missing.
There are many forums reporting similar issues none of which appear to be my scenario.
I enclose the sip debug info and my inbound routing screens
Licence is valid;
SipDebugInfo: License, Valid 1, Available 2, Consumed 0
my avaya is rejecting the call
308529mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
308529mS SipDebugInfo: Present Call, no match from URI in Request Line
308529mS SipDebugInfo: Sending code 404 to method INVITE
I have a static nat supplied throug a juniper netscreen firewall which is currently forwarding all ports to/from the ip office (on 192.168.40.50 to its public ip address)
Call is being directed to my sip user account (if registered) then my IP from an Asterisk PBX as a TRUNK.
asterisk config is;
[barwell_spain]
type=xxx
username=usnerma
secret=password
context=user
qualify=yes
defaultip=213.139.23.182
host=dynamic
Basically
a) Why is my ip office rejecting the call ?
b) why wont my IPO register on the sip server? (have tried register - i do not require STUN) but have set the IP to start STUN anyway, quite often the details change once I have set them.
At this stage i'm only concerned with inbound calls NOT outbound.
Running 4.1(12)
sip-line.jpg
sip-uri.jpg
There are many forums reporting similar issues none of which appear to be my scenario.
I enclose the sip debug info and my inbound routing screens
Licence is valid;
SipDebugInfo: License, Valid 1, Available 2, Consumed 0
my avaya is rejecting the call
308529mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
308529mS SipDebugInfo: Present Call, no match from URI in Request Line
308529mS SipDebugInfo: Sending code 404 to method INVITE
I have a static nat supplied throug a juniper netscreen firewall which is currently forwarding all ports to/from the ip office (on 192.168.40.50 to its public ip address)
Call is being directed to my sip user account (if registered) then my IP from an Asterisk PBX as a TRUNK.
asterisk config is;
[barwell_spain]
type=xxx
username=usnerma
secret=password
context=user
qualify=yes
defaultip=213.139.23.182
host=dynamic
Basically
a) Why is my ip office rejecting the call ?
b) why wont my IPO register on the sip server? (have tried register - i do not require STUN) but have set the IP to start STUN anyway, quite often the details change once I have set them.
At this stage i'm only concerned with inbound calls NOT outbound.
Running 4.1(12)
308512mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
INVITE sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jul 2008 13:12:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 53935 53935 IN IP4 85.13.242.9
s=session
c=IN IP4 85.13.242.9
t=0 0
m=audio 9612 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
308513mS SIP Trunk: 10:Rx
INVITE sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jul 2008 13:12:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 53935 53935 IN IP4 85.13.242.9
s=session
c=IN IP4 85.13.242.9
t=0 0
m=audio 9612 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
308513mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
308520mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
308520mS SipDebugInfo: SIP: ProcessInbound Message
308520mS SipDebugInfo: Create Incoming EndPoint voip
308521mS SipDebugInfo: License, Valid 1, Available 2, Consumed 0
308521mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
308522mS SipDebugInfo: ExtractCallIdFrom Message: Call Id value 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
308522mS SipDebugInfo: ExtractCallerFromMessage: From Tag is as658ccd71
308523mS SipDebugInfo: No Accept Header
308524mS SipDebugInfo: Sending code 100 to method INVITE
308524mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 0
308525mS SipDebugInfo: Sip_sendToNetwork packet of length 352
308526mS SipDebugInfo: SIP Line (10): SendToTarget 550df209, 5060
308526mS SIP Trunk: 10:Tx
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
308526mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
308527mS SipDebugInfo: initialising mTxnContext
308527mS SipDebugInfo: Present call entered
308528mS SipDebugInfo: INVITE Received ep fed54818, dialog fed14dcc
308528mS SipDebugInfo: ExtractRouteFromRecord, entered
308528mS SipDebugInfo: *********************************************************
308528mS SipDebugInfo: State Transtion form Old State 0 to New state 10
308528mS SipDebugInfo: *********************************************************
308529mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
308529mS SipDebugInfo: Present Call, no match from URI in Request Line
308529mS SipDebugInfo: Sending code 404 to method INVITE
308530mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 1
308531mS SipDebugInfo: Sip_sendToNetwork packet of length 355
308531mS SipDebugInfo: SIP Line (10): SendToTarget 550df209, 5060
308531mS SIP Trunk: 10:Tx
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
308532mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
308533mS SipDebugInfo: *********************************************************
308533mS SipDebugInfo: State Transtion form Old State 10 to New state 40
308533mS SipDebugInfo: *********************************************************
308534mS SipDebugInfo: SIP Line (10): Freed Txn Key 2016
308606mS RES: Tue 8/7/2008 13:59:59 FreeMem=47249756(34) CMMsg=4 (5) Buff=100 596 499 1052 3 Links=591
308623mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
ACK sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
308624mS SIP Trunk: 10:Rx
ACK sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
308624mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
308628mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
308628mS SipDebugInfo: SIP: ProcessInbound Message
308628mS SipDebugInfo: Find End Point 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
308629mS SipDebugInfo: Process SIP request dialog fed14dcc, method ACK in Call state 40, Sdp state 0
308631mS SipDebugInfo: State is 40
308632mS SipDebugInfo: SIP Line (10): Cannot free Txn Key 2015
309631mS SipDebugInfo: EPTerminationTimeout, about to delete endpoint
309631mS SipDebugInfo: SIPDialog destructor ... fed14dcc
309632mS SipDebugInfo: Dialog destructor is deleting store d SIP message
309633mS SipDebugInfo: ~SipTrunkEndpoint
309634mS CMTARGET: 0.1018.2 -1 BaseEP: ~CMTargetHandler
313627mS SipDebugInfo: Timer 9 callback
314105mS RES: Tue 8/7/2008 14:00:04 FreeMem=47282144(20) CMMsg=4 (5) Buff=100 596 499 1052 3 Links=622
315059mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
INVITE sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jul 2008 13:12:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 53935 53935 IN IP4 85.13.242.9
s=session
c=IN IP4 85.13.242.9
t=0 0
m=audio 20364 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
315059mS SIP Trunk: 10:Rx
INVITE sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jul 2008 13:12:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 53935 53935 IN IP4 85.13.242.9
s=session
c=IN IP4 85.13.242.9
t=0 0
m=audio 20364 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
315060mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
315066mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
315067mS SipDebugInfo: SIP: ProcessInbound Message
315067mS SipDebugInfo: Create Incoming EndPoint voip
315068mS SipDebugInfo: License, Valid 1, Available 2, Consumed 0
315068mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
315069mS SipDebugInfo: ExtractCallIdFrom Message: Call Id value 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
315069mS SipDebugInfo: ExtractCallerFromMessage: From Tag is as52712853
315069mS SipDebugInfo: No Accept Header
315071mS SipDebugInfo: Sending code 100 to method INVITE
315071mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 0
315072mS SipDebugInfo: Sip_sendToNetwork packet of length 352
315072mS SipDebugInfo: SIP Line (10): SendToTarget 550df209, 5060
315072mS SIP Trunk: 10:Tx
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
315073mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
315073mS SipDebugInfo: initialising mTxnContext
315074mS SipDebugInfo: Present call entered
315074mS SipDebugInfo: INVITE Received ep fed547ec, dialog fed14dcc
315074mS SipDebugInfo: ExtractRouteFromRecord, entered
315075mS SipDebugInfo: *********************************************************
315075mS SipDebugInfo: State Transtion form Old State 0 to New state 10
315075mS SipDebugInfo: *********************************************************
315076mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
315076mS SipDebugInfo: Present Call, no match from URI in Request Line
315076mS SipDebugInfo: Sending code 404 to method INVITE
315076mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 1
315078mS SipDebugInfo: Sip_sendToNetwork packet of length 355
315078mS SipDebugInfo: SIP Line (10): SendToTarget 550df209, 5060
315078mS SIP Trunk: 10:Tx
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
315078mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
315079mS SipDebugInfo: *********************************************************
315080mS SipDebugInfo: State Transtion form Old State 10 to New state 40
315080mS SipDebugInfo: *********************************************************
315080mS SipDebugInfo: SIP Line (10): Freed Txn Key 2016
315169mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
ACK sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
315169mS SIP Trunk: 10:Rx
ACK sip:442033551110@213.139.23.182 SIP/2.0
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
Contact: <sip:08458620019@85.13.242.9>
Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
315170mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget
315173mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1
315173mS SipDebugInfo: SIP: ProcessInbound Message
315174mS SipDebugInfo: Find End Point 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
315175mS SipDebugInfo: Process SIP request dialog fed14dcc, method ACK in Call state 40, Sdp state 0
315177mS SipDebugInfo: State is 40
315177mS SipDebugInfo: SIP Line (10): Cannot free Txn Key 2015
316178mS SipDebugInfo: EPTerminationTimeout, about to delete endpoint
316178mS SipDebugInfo: SIPDialog destructor ... fed14dcc
316178mS SipDebugInfo: Dialog destructor is deleting store d SIP message
316179mS SipDebugInfo: ~SipTrunkEndpoint
sip-line.jpg
sip-uri.jpg
ASKER
Hi,
Further to shepimports reply (thanks), it doesn't really help me any further (sorry!)
sip:442033551110@213.139.2
the IPO is receiving the SIP INVITE but rejecting the message as it doesn't accept the incoming number.
- I think your comment confirms that, but I was sure of that before!
I dont know where in the IPO other than inbound routes how to make it accept the number.
on PSTN trunks you simply add an inbound route with no DID number with the corresponding line group (in my instance this is line:10) and it would route all calls regardless of DID/DDI presented to the pre-defined destination, but i think thats not the case for sip trunks.
I have tried what appears to be every combination in my inbound route without any success. The log is reporting a SIP URI issue and on IPO that appears to be an attribute of the line not the inbound route.
currently have
line 10
incoming number: 442033551110
line 10
incoming number: 442033551110@213.139.23.18
line 10
incoming number: sip:442033551110@213.139.2
What i'm after really is to know exactly whats missing or an example of a working sip config, so i can investigate and formulate my solution from it.
sorry your comment didn't solve my problem, but hope the comments above assist in narrowing down the issue
Thanks
Further to shepimports reply (thanks), it doesn't really help me any further (sorry!)
sip:442033551110@213.139.2
the IPO is receiving the SIP INVITE but rejecting the message as it doesn't accept the incoming number.
- I think your comment confirms that, but I was sure of that before!
I dont know where in the IPO other than inbound routes how to make it accept the number.
on PSTN trunks you simply add an inbound route with no DID number with the corresponding line group (in my instance this is line:10) and it would route all calls regardless of DID/DDI presented to the pre-defined destination, but i think thats not the case for sip trunks.
I have tried what appears to be every combination in my inbound route without any success. The log is reporting a SIP URI issue and on IPO that appears to be an attribute of the line not the inbound route.
currently have
line 10
incoming number: 442033551110
line 10
incoming number: 442033551110@213.139.23.18
line 10
incoming number: sip:442033551110@213.139.2
What i'm after really is to know exactly whats missing or an example of a working sip config, so i can investigate and formulate my solution from it.
sorry your comment didn't solve my problem, but hope the comments above assist in narrowing down the issue
Thanks
ASKER
extract from help;
SIP Calls
For SIP calls, the following fields are used for call matching:
Line Group ID
This field is matched against the Incoming Group settings of the SIP URI (Line | SIP URI). This must be an exact match.
Incoming Number
This field can be used to match the called details (TO) in the SIP header of incoming calls. It can contain a number, SIP URI or Tel URI. For SIP URI's the domain part of the URI is removed before matching by incoming call routing occurs. For example, for the SIP URI mysip@sipitsp.com , only the user part of the URI, ie. mysip, is used for matching.
Incoming CLI
This field can be used to match the calling details (FROM) in the SDP header of incoming SIP calls. It can contain a number, SIP URI, Tel URI or IP address received with SIP calls. For all types of incoming CLI except IP addresses a partial entry can be used to achieve the match, entries being read from left to right. For IP addresses only full entry matching is supported.
I'm assuming therefore that
line 10
incoming number: 442033551110
should be correct.
SIP Calls
For SIP calls, the following fields are used for call matching:
Line Group ID
This field is matched against the Incoming Group settings of the SIP URI (Line | SIP URI). This must be an exact match.
Incoming Number
This field can be used to match the called details (TO) in the SIP header of incoming calls. It can contain a number, SIP URI or Tel URI. For SIP URI's the domain part of the URI is removed before matching by incoming call routing occurs. For example, for the SIP URI mysip@sipitsp.com , only the user part of the URI, ie. mysip, is used for matching.
Incoming CLI
This field can be used to match the calling details (FROM) in the SDP header of incoming SIP calls. It can contain a number, SIP URI, Tel URI or IP address received with SIP calls. For all types of incoming CLI except IP addresses a partial entry can be used to achieve the match, entries being read from left to right. For IP addresses only full entry matching is supported.
I'm assuming therefore that
line 10
incoming number: 442033551110
should be correct.
ASKER
my thoughts at the moment are;
the URI presented by my sip provider is my ip address, and that is not the domain name or ip set in the sip trunk page, so i'm thinking its that - however i have tried to make a false trunk with my ip address instead of the providers details but it doesn't appear to make any difference.
The provider doesn't use STUN so i've tried switching that off, but am still not entirely sure where the problem lies.
the URI presented by my sip provider is my ip address, and that is not the domain name or ip set in the sip trunk page, so i'm thinking its that - however i have tried to make a false trunk with my ip address instead of the providers details but it doesn't appear to make any difference.
The provider doesn't use STUN so i've tried switching that off, but am still not entirely sure where the problem lies.
ASKER CERTIFIED SOLUTION
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I answered my own question, what do I do?
Post a request in the Community Support Zone asking for a refund, and asking the Moderators to close the question; be sure to post the URL to your question. You will be required to post your solution in your original question. A Moderator will post a notice of your request which will give the participants four days to object to the refund. Note that if it resembles one of the suggested comments, the likelihood is that your request will not be granted, but rather, the points will be awarded to the Expert who makes the suggestion.
INVITE sip:442033551110@213.139.23.182 SIP/2.0 **************************
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9
From: "08458620019" ;tag=as658ccd71
To: 442033551110@213.139.23.18
Contact:
Call-ID: 5b0dd2f97269b6e7538dac570a
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jul 2008 13:12:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284
****Skip a few lines****
308529mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header. ******You than see the number trying to be reached
308529mS SipDebugInfo: Present Call, no match from URI in Request Line
308529mS SipDebugInfo: Sending code 404 to method INVITE
308530mS SipDebugInfo: SendSIPResponse, Number of Tag Count, 1
308531mS SipDebugInfo: Sip_sendToNetwork packet of length 355
308531mS SipDebugInfo: SIP Line (10): SendToTarget 550df209, 5060
308531mS SIP Trunk: 10:Tx
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9
From: "08458620019" ;tag=as658ccd71
To: ;tag=106c369239e9cb5c
Call-ID: 5b0dd2f97269b6e7538dac570a
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0
****So basically it is a routing table issue... you need to either add or strip digitz to have to two having a working dial plan... if you dial 442033551110 from an extenson does it work? Does it strip or add digitz? The IPO may require it to be dial in E.164 which would have a format of +442033551110 ... try dialing an internal extension (you should make sure the asterisk extensions are diff than the ones on the IPO to avoid routing issues....
Good Luck