IPO 9.1 with SIP Trunk Problem

[gscadmin]

Hey All,

We installed a new IPOffice 9.1 with a SIP Trunk from (Level 3). Currently we are unable to receive incoming calls. Outgoing works fine. Unfortunately there is not an updated document on the carrier we are using and 9.1.

When we run monitor we can capture the incoming call but it never rings the phone or hits the trunk in system status. I know there can be a lot of variables at this point.

Any help would be appreciated.

(I cleaned up the ip address and numbers below using " X " )

If it matters the 925 area code is calling the 415 area code.

Here is an excerpt from the capture:

16:47:47 332157mS SIP Rx: UDP X.X.X.225:5060 -> X.X.X.70:5060
INVITE sip:+1415XXXXXXX@X.X.X.70:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.225:5060;branch=z9hG4bK06Bd76c4ee844679df4
From: "ABC INC " <sip:+1925XXXXXXX@X.X.X.225:5060>;tag=gK0645cd4c
To: <sip:+1415XXXXXXX@X.X.X.70:5060>
Call-ID: 1829163312_89770089@X.X.X.225
CSeq: 9829 INVITE
Max-Forwards: 54
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: "ABC INC " <sip:+1925XXXXXXX@X.X.X.225:5060>
Supported: 100rel
Content-Length: 258
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 29464 30771 IN IP4 X.X.X.225
s=SIP Media Capabilities
c=IN IP4 X.X.X.194
t=0 0
m=audio 21066 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
16:47:47 332160mS Sip: Association found trunk: SIP Line (17)
16:47:47 332161mS CMCallEvt: 0000000000000000 0.1007.0 -1 BaseEP: NEW CMEndpoint f17f8b30 TOTAL NOW=1 CALL_LIST=0
16:47:47 332161mS Sip: SIP Line (17): sip_trunk_config_items 00020001, voip.flags 00000949
16:47:47 332161mS Sip: SIPDialog f17f72a8 created, dialogs 1
16:47:47 332162mS Sip: SIP Line (17): License, Valid 1, Available 10, Consumed 0
16:47:47 332162mS Sip: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
16:47:47 332163mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from X.X.X.225:5060 to X.X.X.225:5060
16:47:47 332163mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from X.X.X.225:5060 to X.X.X.225:5060
16:47:47 332163mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) Cloned
16:47:47 332164mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent_by: X.X.X.225:5060 trunk
16:47:47 332164mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent by transport: SIP/2.0/UDP trunk
16:47:47 332164mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from X.X.X.225:5060 to X.X.X.225:5060
16:47:47 332164mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) SendSIPResponse: INVITE code 100 SENT TO X.X.X.225 5060
16:47:47 332165mS SIP Tx: UDP X.X.X.70:5060 -> X.X.X.225:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.225:5060;branch=z9hG4bK06Bd76c4ee844679df4
From: "ABC INC " <sip:+1925XXXXXXX@X.X.X.225:5060>;tag=gK0645cd4c
Call-ID: 1829163312_89770089@X.X.X.225
CSeq: 9829 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.0.0 build 437
To: <sip:+1415XXXXXXX@X.X.X.70:5060>;tag=25ed53b0f88e417b
Content-Length: 0

16:47:47 332165mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) INVITE Received ep 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk

Endpoint(f17f8b30), dialog f17f72a8
16:47:47 332165mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) UpdateSIPCallState SIPDialog::INITIAL(0) ->

SIPDialog::INVITE_RCVD(9)
16:47:47 332166mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) Present Call, no match (+1415XXXXXXX) from URI in Request Line
16:47:47 332166mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) Present Call, no match (+1415XXXXXXX) from URI in To header or

(+1415XXXXXXX) from request URI
16:47:47 332166mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent_by: X.X.X.225:5060 trunk
16:47:47 332166mS Sip: SIPDialog::ExtractResponseParamsFromViaHeader remote sent by transport: SIP/2.0/UDP trunk
16:47:47 332166mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForResponse from X.X.X.225:5060 to X.X.X.225:5060
16:47:47 332167mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) SendSIPResponse: INVITE code 404 SENT TO X.X.X.225 5060
16:47:47 332167mS SIP Tx: UDP X.X.X.70:5060 -> X.X.X.225:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP X.X.X.225:5060;branch=z9hG4bK06Bd76c4ee844679df4
From: "ABC INC " <sip:+1925XXXXXXX@X.X.X.225:5060>;tag=gK0645cd4c
Call-ID: 1829163312_89770089@X.X.X.225
CSeq: 9829 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.0.0 build 437
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:+1415XXXXXXX@X.X.X.70:5060>;tag=25ed53b0f88e417b
Content-Length: 0

16:47:47 332168mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) UpdateSIPCallState SIPDialog::INVITE_RCVD(9) ->

SIPDialog::FINAL(28)
16:47:47 332484mS RES: Wed 18/2/2015 16:47:48 FreeMem=56898896 (Heap=56723760(3) + Cache=175136) MemObjs=7541(Max 7919) CMMsg=5(6) Buff=5200

1362 1000 7444 5 Links=51353(58000) BTree=14661 CPU=04.45% CPUStats=04.28%/1/3/5766/21527/23709/00.00%/0/02.4
1% MCR=0 MCW=0
16:47:47 332484mS RES2: IP 500 V2 9.1.0.0 build 437 Tasks=50 RTEngine=0 CMRTEngine=0 ExRTEngine=0 Timer=56 Poll=0 Ready=0 CMReady=0 CMQueue=0

VPNNQueue=0 Monitor=1 SSA=1 TCP=22(TLS=5) TAPI=0 ASC=1 SYS=MNTD OPT=UMNT SDSPD=2034
16:47:47 332484mS RES4: XML MemObjs=63 PoolMem=4876404(2) FreeMem=4861100(0)
16:47:47 332485mS RES5: CLog MemObjs=7 FreePoolMem(Objs)=4836(93) TotalMem=85350
16:47:47 332759mS PRN: LoadCountControl: Marked As Successful Boot2
16:47:49 334169mS SIP Tx: UDP X.X.X.70:5060 -> X.X.X.225:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP X.X.X.225:5060;branch=z9hG4bK06Bd76c4ee844679df4
From: "ABC INC " <sip:+1925XXXXXXX@X.X.X.225:5060>;tag=gK0645cd4c
Call-ID: 1829163312_89770089@X.X.X.225
CSeq: 9829 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
Server: IP Office 9.1.0.0 build 437
Reason: Q.850;cause=1;text="Unallocated (unassigned) number"
To: <sip:+1415XXXXXXX@X.X.X.70:5060>;tag=25ed53b0f88e417b
Content-Length: 0

16:47:49 334177mS SIP Rx: UDP X.X.X.225:5060 -> X.X.X.70:5060
ACK sip:+1415XXXXXXX@X.X.X.70:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.225:5060;branch=z9hG4bK06Bd76c4ee844679df4
From: "ABC INC " <sip:+1925XXXXXXX@X.X.X.225:5060>;tag=gK0645cd4c
To: <sip:+1415XXXXXXX@X.X.X.70:5060>;tag=25ed53b0f88e417b
Call-ID: 1829163312_89770089@X.X.X.225
CSeq: 9829 ACK
Max-Forwards: 70
Content-Length: 0

16:47:49 334178mS Sip: Find End Point2 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint (f17f8b30) Sip CallId 1829163312_89770089@X.X.X.225
16:47:49 334179mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) Process SIP request dialog f17f72a8, method ACK in state

SIPDialog::FINAL(28)
16:47:50 335179mS Sip: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint(f17f72a8) TerminationTimeout, dialog is in SIPDialog::FINAL
16:47:50 335179mS Sip: SIPDialog f17f72a8 deleted, dialogs 0
16:47:50 335179mS CMCallEvt: 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint: StateChange: END=X CMCSIdle->CMCSDelete
16:47:50 335179mS Sip: ~SipTrunkEndpoint 0a0a01fa000003ef 17.1007.1 -1 SIPTrunk Endpoint
16:47:50 335180mS CMTARGET: 0a0a01fa000003ef 17.1007.1 -1 BaseEP: ~CMTargetHandler f17fbdc0 ep f17f8b30
16:47:50 335180mS CMCallEvt: 0a0a01fa000003ef 17.1007.1 -1 BaseEP: DELETE CMEndpoint f17f8b30 TOTAL NOW=0 CALL_LIST=0
16:47:55 339178mS Sip: sip_indicateTimeOut Timer 9
16:47:55 339178mS Sip: Timer 9 callback didn't find dialog, method INVITE, callid 1829163312_89770089@X.X.X.225
16:47:55 339420mS RES: Wed 18/2/2015 16:47:55 FreeMem=56974896 (Heap=56756688(3) + Cache=218208) MemObjs=7223(Max 7919) CMMsg=5(6) Buff=5200

1362 1000 7443 5 Links=51332(58000) BTree=14661 CPU=11.60% CPUStats=06.00%/1/3/5766/21482/23709/00.04%/0/02.4
0% MCR=0 MCW=0
16:47:55 339421mS RES2: IP 500 V2 9.1.0.0 build 437 Tasks=49 RTEngine=0 CMRTEngine=0 ExRTEngine=0 Timer=56 Poll=0 Ready=0 CMReady=0 CMQueue=0

VPNNQueue=0 Monitor=1 SSA=1 TCP=22(TLS=5) TAPI=0 ASC=1 SYS=MNTD OPT=UMNT SDSPD=2034
16:47:55 339421mS RES4: XML MemObjs=63 PoolMem=4876404(2) FreeMem=4861100(0)
16:47:55 339421mS RES5: CLog MemObjs=7 FreePoolMem(Objs)=4836(93) TotalMem=85350
16:47:57 341944mS Sip: SIP Line (17): Options timer expired
16:47:57 341944mS Sip: SIP Line (17): Setting options timer 30 seconds
16:47:57 341944mS Sip: SIPDialog f17eeab8 created, dialogs 1
16:47:57 341945mS Sip: (f17eeab8) OPTIONS SENT TO X.X.X.225 5060
16:47:57 341946mS SIP Tx: UDP X.X.X.70:5060 -> X.X.X.225:5060
OPTIONS sip:X.X.X.225 SIP/2.0
Via: SIP/2.0/UDP X.X.X.70:5060;rport;branch=z9hG4bK95faa720663040ba7d7dcaf995dbfbf8
From: <sip:X.X.X.70>;tag=3bd122b1b75d71ad
To: <sip:X.X.X.225>
Call-ID: e907e7abac30d53fc13ff8c5a2a7b037
CSeq: 1859294718 OPTIONS
Contact: <sip:X.X.X.70:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 9.1.0.0 build 437
Content-Length: 0

16:47:57 341953mS SIP Rx: UDP X.X.X.225:5060 -> X.X.X.70:5060
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP X.X.X.70:5060;branch=z9hG4bK95faa720663040ba7d7dcaf995dbfbf8;rport=5060
From: <sip:X.X.X.70>;tag=3bd122b1b75d71ad
To: <sip:X.X.X.225>
Call-ID: e907e7abac30d53fc13ff8c5a2a7b037
CSeq: 1859294718 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE



Thank you,
gscadmin

 

[esoderholm]

Check your incoming Call route, seems the IPO cant find any matching incoming number.

E Soderholm,

Aut inveniam viam aut faciam.

 

[tlpeter]

I see a 404 which means the call is actually hitting the IPO but is not recognized.
Mostly this is a problem with no URI in the system with the correct format of the number.

BAZINGA!

I'm not insane, my mother had me tested!

 

[cnesstech]

Peter, Level3 is sending us e.164 numbering format so it shows up as +1 10 digits. I'm having trouble getting anything in the ICR or URI fields to allow a match. Any suggestions? Thanks.

 

[amriddle01]

Make a *** URI with a unique outgoing line group, incoming group 0, then just add the DDI to incoming call route

 

[gscadmin]

Thanks everyone, we got it working.

 

[tlpeter]

OK, that is nice.
What AMriddle said is using wildcards which means and URI.
If there is no * on all fields then the system look for the URI -> +1123456789 (example)
If it is not there then you will get a 404.
The incoming callroute must have the number but does not give a 404.


BAZINGA!

I'm not insane, my mother had me tested!