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issue with ipo sip not working correct

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ntfd1019

Instructor
Apr 14, 2016
1
US
hello all, newbie here for sip on ipo, i can get ipo to make sip calls with out a hitch, incoming calls are another thing, I have been working on this for some time and im coming up with a call not matched but it is clearly in the icr any suggections


20:20:58 66502mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) INVITE Received ep 17.1006.1 -1 SIPTrunk Endpoint(f51d6a68), dialog f51d578c
20:20:58 66502mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) LR is On and route is Route: <sip:65.254.44.194;lr=on;ftag=gK081887af;vsf=AAAAABwFBwwGAA4LDwV3AHcYBR0DHQAAAB8BMTA-;dlgcor=8b9.8ae>
20:20:58 66503mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(9)
20:20:58 66503mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) Present Call, no match (5282958535) from URI in Request Line
20:20:58 66503mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) Present Call, no match (+17408681914) from URI in To header or (5282958535) from request URI
20:20:58 66503mS Sip: SIPDialog::ExtractPortFromViaHeader remote: 65.254.44.194 trunk
20:20:58 66503mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) SendSIPResponse: rport not found - 0
20:20:58 66504mS Sip: 17.1006.1 -1 SIPTrunk Endpoint(f51d578c) SendSIPResponse: INVITE code 404 SENT TO 65.254.44.194 5060



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 http://files.engineering.com/getfile.aspx?folder=8206e457-1a96-4e28-9795-d9e9909e7d8a&file=Capture2.PNG
It would be usefull to add the full trace taken by monitor with the default filter and additional the SIP filter enabled.
The shown trace is not enough to go on.
 
It doesn't check ICR unless you add a *** URI, and that number isn't a URI so no match is possible :)

 
Try adding the full number to the SIP tab of the user.

SIP Name and contact to be the extension number you are trying to call.

Give it a go
 
Forgot to mention

If the calling party is sending 10 digits then the SIP entry needs to contain the 10 digits even if the called party is configured with lesser digits

i.e
Calling party calls 011 555 555
Extension /user = 555

Sip entry need to be configured with 011555 555.

This worked for
 
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