Adding SIP Trunk to 11.0 get 404 Not Found

[rcc1000]

Hello All,
Installing new SIP Trunk & can't get incoming calls to the providers test numbers. Have the licenses installed & say they are valid.
The provider is a small independent provider & I never configured a system using them before.
Haven't even programmed very many SIP Trunks at all, so may be dumb about this.

When trying to call in, from Monitor, get the 404 Not Found & Reason "Unallocated (unassigned) number"
Then in System Status get alarm under Configuration "line had no Incoming Call Route for Call"

I have the number trying to call in the Incoming Call Route, even deleted a few times & put back in.
Then even deleted my whole SIP Trunk & put back in, no luck.

I'm trying to copy the Call Details/URI from another smaller provider that I have working for a different system & thinking that may be where I have something not working with this new provider.

Have two different URIs, 1 with Incoming 101/Outgoing 199 Group numbers, matching my Credentials name, Local URI/Contact (the same for both) Display=Auto, Content=Auto, Outgoing/Incoming Calls=Caller & Forwarding/Twinning=Original Caller

Second URI I have as Incoming/Outgoing Group & Credentials same as the first URI, Local URI/Contact (the same for both) Display=Use Internal Data, Content=Use Internal Data, Outgoing/Incoming Calls=Caller, Forwarding/Twinning=Original Caller
Then have Diversion Header checked with Display/Content=Use Internal Data, Outgoing calls & Forwarding/Twinning=Caller finally Incoming Calls=none & cannot change this from the drop box.

In the Incoming Call Route I have the correct Line Group ID 101, Incoming number=the test number the provider gave me &
changed destination to numerous users or the Auto Attendant.

Me & the provider tech support been fighting this for a while, any help/questions please respond supposed to cut the customer over to this tomorrow.
Thank You everyone

 

[IPGuru]

for starters leave the incoming number field blank
it should then match any incomming number & route to the destination

once that works you know the trunck config is ok & you can start trying to sort out the DDI routing (check the format of the number presented)


Do things on the cheap & it will cost you dear

ACSS

 

[rcc1000]

Thanks IPGURU,
I cleared out the Incoming number field and now the calls will route to my AA.

The next step is to get DID working. I added the incoming 4 digits to the SIP tab on the User and created an Incoming call route for those same 4 digits to that user. The AA still answered the call.

I am sure the CO is sending 4 digits from this capture:

INVITE sip:niacog@159.242.43.49:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.83.31.225:5060;branch=z9hG4bK+812cc1e752e3c3caf5ce927c081a61441+sip+1+b932e517
From: "CLEAR LAKE IND" <sip:6413572462@172.83.31.225:5060>;tag=172.83.31.225+1+a260ac81+c3038528
To: <sip:5345@159.242.43.49>
CSeq: 120379184 INVITE
Expires: 180
Content-Length: 187
Call-Info: <sip:172.83.31.225:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Supported: resource-priority,siprec, 100rel
Contact: <sip:1b50bfd6740c0ff2f80f4f8c84ea9b26@172.83.31.225:5060>;isup-oli=00
Content-Type: application/sdp
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name, ua-profile
Call-ID: 0gQAAC8WAAACBAAALxYAANhhxuwp59blb/GmMpOgDQ7dsHpt2lUPfEFDZ7SEU/Ox@172.83.31.225
Organization: MetaSwitch
Max-Forwards: 69
P-Asserted-Identity: "CLEAR LAKE IND" <sip:6413572462@10.3.0.87:5060>
Accept: application/sdp, application/dtmf-relay

v=0
o=- 70598188794145 70598188794145 IN IP4 172.83.31.225
s=-
c=IN IP4 172.83.31.225
t=0 0
m=audio 39516 RTP/AVP 0 2 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=ptime:20