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IP500v2 9.1.7, Incoming sip calls failing with Incompatible Destination, but ICR is setup

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DaveElder

DaveElder

MIS
Jun 18, 2010
2
FR
I've been fighting this all weekend with no joy and am hoping someone can help.

Had a IP0500v1 chassis with 8.1, replaced chassis with a v2, added new licences, recreating config from scratch.
Sip trunks added, SIP URI set to Use Internal Data. full DDI number added to SIP tab of users. ICR set for full DDI & pointing to user extension.
If I have a default ICR then calls come in to that Group. If I don't have a default ICR then all incoming calls fail with an error

The following line had no Incoming Call Route for a call.
Line: 18
Line Group ID: 5
DDI: 44------5320

ICR is set to that line group and DDI number.

Monitor trace makes it look like it matches the ICR, but fails with Incompatible Destination.
Ext 320 is the destination if the ICR.It is a 9650C IP handset, but same problem if I call a DS 5420
We use Orbis as SIP provider (Gamma reseller). Outgoing calls are ok.

Is there something I've missed that is new to 9.1?
Thanks
Dave


13:57:51 1522399mS SIP Call Rx: 18
INVITE sip:Company@nn.nn.nn.157:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 91.195.228.171:5060;branch=z9hG4bK-524287-1---99cc884a73b9c917;rport
Via: SIP/2.0/UDP 91.195.228.171:5061;branch=z9hG4bK-5nu35tos2pkqhvsz;rport=5061
Max-Forwards: 69
Record-Route: <sip:91.195.228.171:5060;lr;transport=UDP>
Contact: "Anonymous"<sip:91.195.228.171:5061>
To: <sip:44------5320@91.195.228.171>
From: <sip:-----277@91.195.228.171>;tag=rhvjdco63vvgvezl.o
Call-ID: 27138001-3680168272-614696@MSX18.gammatelecom.com
CSeq: 374 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
P-Asserted-Identity: <sip:-------277@91.195.228.171>
Remote-Party-ID: <sip:-------277@91.195.228.171>;party=calling
h323-conf-id: 2329202124-1645941222-2181198982-2060463932
Portasip-3264-action: offer 1
cisco-GUID: 2329202124-1645941222-2181198982-2060463932
Content-Length: 264

v=0
o=Sippy 3192036746687722048 0 IN IP4 91.195.228.171
s=sip call
t=0 0
m=audio 36168 RTP/AVP 0 18 101
c=IN IP4 91.195.228.171
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
13:57:51 1522399mS CMCallEvt: 0000000000000000 0.1053.0 -1 BaseEP: NEW CMEndpoint f196b918 TOTAL NOW=1 CALL_LIST=0
13:57:51 1522399mS Sip: SIP Line (18): sip_trunk_config_items 00020001, voip.flags 00000948
13:57:51 1522400mS Sip: SIPDialog f196a06c created, dialogs 1 txn_keys 1
13:57:51 1522400mS Stun: Info: Line 18: Not using STUN for media in this case.
13:57:51 1522401mS Sip: SIP Line (18): License, Valid 1, Available 20, Consumed 0
13:57:51 1522401mS Sip: SIPTrunkEndpointDialogOwner::SetRemoteAddressForRequest from 91.195.228.171:5060 to 91.195.228.171:5061
13:57:51 1522403mS SIP Call Tx: 18
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.195.228.171:5060;branch=z9hG4bK-524287-1---99cc884a73b9c917;rport
Via: SIP/2.0/UDP 91.195.228.171:5061;branch=z9hG4bK-5nu35tos2pkqhvsz;rport=5061
Record-Route: <sip:91.195.228.171:5060;lr;transport=UDP>
From: <sip:------277@91.195.228.171>;tag=rhvjdco63vvgvezl.o
Call-ID: 27138001-3680168272-614696@MSX18.gammatelecom.com
CSeq: 374 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer
Server: IP Office 9.1.7.0 build 163
To: <sip:44------5320@91.195.228.171>;tag=747eb902990d774b
Content-Length: 0

CMSetup
Line: type=SIPLine 18 Call: lid=18 id=1053 in=1
Called[320] Type=Default (100) Reason=CMDRdirect SndComp Calling[--------277@91.195.228.171] Type=Unknown Plan=Default
BC: CMTC=Speech CMTM=Circuit CMTR=64 CMST=Default CMU1=ALaw
IE CMIEFastStartInfoData (6) 4 item(s)
IE CMIECallingPartyName (110)(Type=CMNameDefault) name=Anonymous
IE CMIERespondingPartyName (228)(Type=CMNameDefault) name=Anonymous
IE CMIERespondingPartyNumber (230)(P:100 S:100 T:0 N:100 R:4) number=------277@91.195.228.171
IE CMIEDeviceDetail (231) c0a8010a0000041d LOCALE=eng HW=15 VER=9 class=CMDeviceSIPTrunk type=0 number=18 channel=1 features=0x0 rx_gain=32 tx_gain=32 ep_callid=1053 ipaddr=192.168.1.10 apps=0 loc=999 em_loc=999 features2=0x0 is_spcall=1
IE CMIEDIDNumber (245)(P:100 S:100 T:100 N:100 R:4) number=Company

13:57:51 1522410mS CD: CALL: 18.1053.1 BState=Idle Cut=1 Music=0.0 Aend="Line 18" (0.0) Bend="" [] (0.0) CalledNum=320 () CallingNum=--------277@91.195.228.171 (Anonymous) Internal=0 Time=3 AState=Idle

13:57:51 1522410mS CMCallEvt: c0a8010a0000041d 18.1053.1 12 SIPTrunk Endpoint: StateChange: END=A CMCSIdle->CMCSDialInitiated
13:57:51 1522410mS CMTARGET: c0a8010a0000041d 18.1053.1 12 SIPTrunk Endpoint: LOOKUP CALL ROUTE: GID=5 type=100 called_party=320 sub= calling=--------277@91.195.228.171 calling_sub= dir=in complete=1 ses=0
13:57:51 1522413mS CMLOGGING: CALL:2016/08/1413:57,00:00:00,000,--------277@91.195.228.171,I,320,320,Anonymous,,,0,,"",0,n/a
13:57:51 1522413mS CD: CALL: 18.1053.1 BState=Idle Cut=0 Music=0.0 Aend="Line 18" (0.0) Bend="" [] (0.0) CalledNum=320 () CallingNum=--------277@91.195.228.171 (Anonymous) Internal=0 Time=6 AState=Dialling
13:57:51 1522413mS CD: CALL: 18.1053.1 Deleted
13:57:51 1522413mS CMLineTx: v=0
CMReleaseComp
Line: type=SIPLine 18 Call: lid=18 id=1053 in=1
Cause=88, Incompatible destination
13:57:51 1522413mS Sip: c0a8010a0000041d 18.1053.1 -1 SIPTrunk Endpoint(f196b918) received CMReleaseComp
13:57:51 1522414mS SIP Call Tx: 18
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 91.195.228.171:5060;branch=z9hG4bK-524287-1---99cc884a73b9c917;rport
Via: SIP/2.0/UDP 91.195.228.171:5061;branch=z9hG4bK-5nu35tos2pkqhvsz;rport=5061
Record-Route: <sip:91.195.228.171:5060;lr;transport=UDP>
From: <sip:--------277@91.195.228.171>;tag=rhvjdco63vvgvezl.o
Call-ID: 27138001-3680168272-614696@MSX18.gammatelecom.com
CSeq: 374 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,NOTIFY
Supported: timer
Server: IP Office 9.1.7.0 build 163
Reason: Q.850;cause=88;text="Incompatible destination"
To: <sip:44-------5320@91.195.228.171>;tag=747eb902990d774b
Content-Length: 0
 
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Have now fixed it.
The new Advanced tab in the SIP line has a Call Routing Method. defaults to Request URI but had to change it to To Header.

Dave
 
Upvote 0 Downvote
Yepp, apparently they are not sending the called number in the Request URI which is most common.

"Trying is the first step to failure..." - Homer
 
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