inbound sip call doesnt connect 1

[baldwincl]

I have an IP500 running 4.2.11 connected to an LGS box. I can call users there no problem but i want them to be able to use my local trunks. My short code 9N dial N works for local phones but i get the following line when calls come in.

3598714mS SipDebugInfo: 19.1070.1 -1 SIPTrunk Endpoint(f55f000c) Present Call, no match (97771600) from URI in To header.


when they call me at ext 301 i get the following

3580245mS SIP Trunk: 19:Rx
BYE sip:Ext301@10.10.100.101:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.20.12.25:5061;branch=z9hG4bK51766fe61961ac7a4e0ead91772673d3204-164640404439
To: "Ext301" <sip:Ext301@10.10.100.101>;tag=0372adab526df68e
From: <sip:1685@10.20.12.25>;tag=204-164619376511
Call-ID: 41337e999c837afe77b10597f2f9aaa5@10.10.100.101
CSeq: 1 BYE
Max-Forwards: 70
User-Agent: SIP-Stack v3.0.3
Content-Length: 0

3580247mS SipDebugInfo: SIPDialog TXN : Decoding of message Succeded 1
3580247mS SipDebugInfo: Find End Point 19.1067.0 21 SIPTrunk Endpoint (f55fee30) Sip CallId 41337e999c837afe77b10597f2f9aaa5@10.10.100.101
3580248mS SipDebugInfo: 19.1067.0 21 SIPTrunk Endpoint(f55fe20c) Process SIP request dialog f55fe20c, method BYE in state SIPDialog::CALL_UP(16)
3580248mS SipDebugInfo: 19.1067.0 21 SIPTrunk Endpoint(f55fe20c) Cannot Clone, message does not exist !!!!!!!
3580248mS SipDebugInfo: 19.1067.0 21 SIPTrunk Endpoint(f55fe20c) SendSIPResponse: BYE SENT TO 10.20.12.25 5060
3580249mS SipDebugInfo: 19.1067.0 21 SIPTrunk Endpoint(f55fe20c) Sending code 200 to method BYE
3580249mS SipDebugInfo: 19.1067.0 21 SIPTrunk Endpoint(f55fe20c) SendSIPResponse, Number of Tag Count, 1
3580249mS SIP Trunk: 19:Tx
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.20.12.25:5061;branch=z9hG4bK51766fe61961ac7a4e0ead91772673d3204-164640404439
From: <sip:1685@10.20.12.25>;tag=204-164619376511
To: "Ext301" <sip:Ext301@10.10.100.101>;tag=0372adab526df68e
Call-ID: 41337e999c837afe77b10597f2f9aaa5@10.10.100.101
CSeq: 1 BYE
Content-Length: 0

3580249mS SIP Tx: UDP 10.10.100.101:5060 -> 10.20.12.25:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 10.20.12.25:5061;branch=z9hG4bK51766fe61961ac7a4e0ead91772673d3204-164640404439
From: <sip:1685@10.20.12.25>;tag=204-164619376511
To: "Ext301" <sip:Ext301@10.10.100.101>;tag=0372adab526df68e
Call-ID: 41337e999c837afe77b10597f2f9aaa5@10.10.100.101
CSeq: 1 BYE
Content-Length: 0

3580251mS SipDebugInfo: 19.1067.0 21 SIPTrunk Endpoint(f55fe20c) UpdateSIPCallState SIPDialog::CALL_UP(16) -> SIPDialog::BYE_RCVD(22)
3580251mS CMLineRx: v=0
CMReleaseComp
Line: type=IPLine 19 Call: lid=19 id=1067 in=0
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=61685
IE CMIEDeviceDetail (231) LOCALE=enu HW=8 VER=4 class=CMDeviceSIPTrunk type=0 number=19 channel=0 rx_gain=32 tx_gain=32 ep_callid=1067 ipaddr=10.10.100.101 apps=0
3580251mS CMCallEvt: 19.1067.0 21 SIPTrunk Endpoint: StateChange: END=B CMCSConnected->CMCSCompleted
3580255mS CMLOGGING: CALL:2009/01/3016:39,00:00:15,005,301,O,1685@10.20.12.25,61685,Ext301,,,0,,""n/a,0
3580256mS CD: CALL: 0.1065.0 BState=Disconnecting Cut=1 Music=0.0 Aend="Ext301(301)" (20.1) Bend="Line 19" [Line 19] (0.0) CalledNum=1685@10.20.12.25 () CallingNum=301 (Ext301) Internal=0 Time=32028 AState=Connected
3580256mS CD: CALL: 0.1065.0 Deleted
3580257mS CMExtnEvt: Ext301: CALL LOST (CMCauseNormal)
3580257mS CMExtnEvt: Ext301: Extn(301) Calling Party Number(301) Type(CMNTypeInternal)

Any help would be appreciated.
Thanks

"So this is how liberty dies. With thunderous applause.

 

[hibroth]

Do you mean you need Sip calls enter the IPO via SIP trunk then go out through your local IPO ISDN trunk ?

 

[Bas1234]

How is your URI setup done?
You'll need to put it on "Use User Data" then on the "User" there you'll find another "SIP" tab change the contact to the number you would like to.

Greetzzz..Bas

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[baldwincl]

hibroth you are correct thats what i am trying.

Bas- URI is set for user data, however since i am trying to make calls out to the PSTN is there a wild card or do i have to put in a user for every number they would want to call?

"So this is how liberty dies. With thunderous applause.

 

[IPOfficeIreland]

i have tried this but failed. it is possible using vmpro however.

 

[hibroth]

I had it, no wildcards are allowed.
That means no way to enter via SIP and generically get access to local trunk.
As TheTaker said the only chance seems using VMPro, but I have never been tested.

 

[IPOfficeIreland]

unless.... thinking aloud

you route the incoming sip call to a fixed DDI number which connects back into the IPO as is set up as a FNE Mobility User number - therefore giving dialtone?

sounds like a lot of resources to use though. I wonder could we give FNE00 to an inbound SIP call as I am sure you could mimic the incoming id for certain calls so that it gets a match?

e.g your other system when looking for a trunk dials a particular id number which sends caller id 1234. 1234 is set up as a mobility user. All ICR on the trunk point to FNE00 for dial tone.

I will check this

 

[IPOfficeIreland]

okay, should be workable but getting a bit stuck on this.

IPO to IPO via SIP. Sending caller id 1234567 from 1 side.

Other side receives call on ICR Any Voice to Short Code *5200 FNE 00 - Dial Tone.

The caller id is also set up as mobility 1234567 on a user so the incoming call should get dial tone.

I have a short code on site a: *88 dial Site B 1234567@ip.

Connects and according to SSA i have Trunk access using the subscriber but no dail tone or can't dial out.
Traces arent' showing anything unusual.

I have now tried an external DDI number on the SIP Site A to ring to Site B and give dial tone, same issue.

Will now try external DDI to Site A or Site B to test that FNE is working for sip (works for isdn, according to doc should be working for sip)

Will continue and post back.

 

[IPOfficeIreland]

okay, working using above set up, so that any person can now ring the DDI and get dial tone through the second site.

For security you can capture caller id and prevent the call.

Provider must lock down dtmf to rfc2833 also.

so this will work for linking 2 pbxs together andneeding dial tone on site a or b via sip.

or even better if you have a large number of mobility users just buy a second ipo.

good luck!

 

[IPOfficeIreland]

do you think it is worth doing up a tek tips help on this?

 

[baldwincl]

If you mean a tip i could do that once it i can put it all together. Stupid work is getting in the way.

Also Thanks for all of the help!

"So this is how liberty dies. With thunderous applause.

 

[IPOfficeIreland]

what i meant if anyone was interested i would put together a simple step by step on how this is achieved. my previous posts were a bit all over the place....