SIP Trunk to CUCM

[Areyouserious]

Hi all.

I'm trying to connect a SIP trunk line to my Cisco call manager.

up until now I've been using Asterisk to connect a few 7940s.

I'm not sure if the sip trunking set up with asterisk is the same with CUCM, as I'm using a Chan_SIP trunk.

I've been having a play around and some googling but no luck.

What is confusing me the most is where exactly in CUCM do you configure the 

SIP Registration string, account number and password/pin credentials to connect to the sip trunk?

Just wondering if anyone could point me in the right direction.

Thanks.

[Aaron Harrison]

Hi

You would configure those things on your SBC. If you want a Cisco SBC, you would use CUBE, which is a licensed software feature that runs on a normal Cisco router.

Regards

Aaron

[Areyouserious]

Thanks Aaron.

So there's no way that you can directly connect the trunk line form the internet strait to the CUCM server?

[Vivek Batra]

CUCM supports SIP Trunk and you can directly connect it to the internet (not preferred) however to be noted that CUCM doesn't support SIP REGISTER on its trunk, it's configured as peer-to-peer trunk. Hence if your ITSP needs SIP REGISTER for SIP trunk and as mentioned by Aaron, you need to have SBC in between for security and registration. CUBE is one of the choice and by the way, can use any vendor SBC in between.

[Areyouserious]

Thanks for your help guys,

I purchased a Cisco 2811 with Ver 12.

Ive been goggling for hours trying to find a set of instructions to configure the CUBE.

I can see that you have to configure a few things such as pointing cucm to the cube IP int, creating routes, and things like that.

Although I have learnt allot in the last 48 hours from hunting around, I'm still confused as to how you actually configure the string registration, password and account username for my SIP account with my provider ect.

Also, not too sure how to point the cube to the sip providers SIP server IP over the internet.

Is anyone able to point me in the right direction?

I thought that perhaps a "realm" might be what I'm looking at configuring but not too sure....

[Aaron Harrison]

There are a million ways to configured a CUBE. One way to auth is like so :http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/99863-cube-config.html#sip

Aaron

[Areyouserious]

Thanks Aaron!

That looks really helpful, ill give it a crack.

Best wishes.

[Areyouserious]

Well.....

Ive been trying for a couple of hours.

So far ive got call manager talking to CUBE.

but no success dialing out

This is the packet debug dump form the cube when ever i make a call.

*Feb 12 15:39:45.727: //-1/8F405F800000/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=2001
   ----- ccCallInfo IE subfields -----
   cisco-ani=2001
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=994467786
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=-1
*Feb 12 15:39:45.727: //-1/8F405F800000/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x45D82B28, Call Info(
   Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=994467786(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=13
*Feb 12 15:39:45.727: //-1/8F405F800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Feb 12 15:39:45.727: //-1/8F405F800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Feb 12 15:39:45.727: //13/8F405F800000/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=994467786(TON=Unknown, NPI=Unknown))
*Feb 12 15:39:45.727: //13/8F405F800000/CCAPI/cc_process_call_setup_ind:
   Event=0x46242640
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCallSetContext:
   Context=0x44D92674
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 13 with tag 1 to app "_ManagedAppProcess_Default"
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=TRUE, Mode=0,
   Outgoing Dial-peer=2, Params=0x44D93324, Progress Indication=NULL(0)
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCheckClipClir:
   In: Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCheckClipClir:
   Out: Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCallSetupRequest:
   Destination Pattern=9........, Called Number=994467786, Digit Strip=FALSE
*Feb 12 15:39:45.731: //13/8F405F800000/CCAPI/ccCallSetupRequest:
   Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=994467786(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=2001, Final Destination Flag=TRUE,
   Guid=8F405F80-0001-0000-0000-00240401A8C0, Outgoing Dial-peer=2
*Feb 12 15:39:45.735: //13/8F405F800000/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=2001
   ----- ccCallInfo IE subfields -----
   cisco-ani=2001
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=1
   dest=994467786
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=-1
*Feb 12 15:39:45.735: //13/8F405F800000/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x45D82B28, Interface Type=3, Destination=, Mode=0x0,
   Call Params(Calling Number=2001(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
   Called Number=994467786(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=2, Call Count On=FALSE,
   Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
*Feb 12 15:39:45.735: //14/8F405F800000/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1
*Feb 12 15:39:45.735: //14/8F405F800000/CCAPI/ccCallSetContext:
   Context=0x44D932D4
*Feb 12 15:39:45.735: //13/8F405F800000/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2
*Feb 12 15:39:45.735: //14/8F405F800000/CCAPI/cc_api_call_proceeding:
   Interface=0x45D82B28, Progress Indication=NULL(0)
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/cc_api_call_disconnected:
   Cause Value=47, Interface=0x45D82B28, Call Id=14
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=47, Retry Count=0)
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/ccCallSetAAA_Accounting:
   Accounting=0, Call Id=14
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/ccCallDisconnect:
   Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=47)
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/ccCallDisconnect:
   Cause Value=47, Call Entry(Responsed=TRUE, Cause Value=47)
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x45D82B28, Tag=0x0, Call Id=14,
   Call Entry(Disconnect Cause=47, Voice Class Cause Code=0, Retry Count=0)
*Feb 12 15:39:45.739: //14/8F405F800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
*Feb 12 15:39:45.743: //13/8F405F800000/CCAPI/ccCallDisconnect:
   Cause Value=47, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Feb 12 15:39:45.743: //13/8F405F800000/CCAPI/ccCallDisconnect:
   Cause Value=47, Call Entry(Responsed=TRUE, Cause Value=47)
*Feb 12 15:39:45.763: //13/8F405F800000/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x45D82B28, Tag=0x0, Call Id=13,
   Call Entry(Disconnect Cause=47, Voice Class Cause Code=0, Retry Count=0)
*Feb 12 15:39:45.763: //13/8F405F800000/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent

---------------------------------------------------------------------------------------------

Might also note that this is my config below

---------------------------------------------------------------------------------------------

Router  Cisco 2811

Building configuration...

Current configuration : 1722 bytes
!
version 12.4
service config
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname TelephoneRouter
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
no network-clock-participate wic 0
!
!
ip cef
!
!
ip name-server 192.168.1.1
!
!
voice-card 0
 no dspfarm
!
!
!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
!
!
!
!
!
!
!
!
!
!
!
!
!
username admin privilege 15 password 0 G689EW7HG98WH6T385JN4
!
!
!
!
!
!
!
interface FastEthernet0/0
 ip address 192.168.1.16 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0
 no ip address
 shutdown
 clock rate 2016000
!
interface Serial0/0/1
 no ip address
 shutdown
 clock rate 2016000
!
interface Serial0/0/2
 no ip address
 shutdown
 clock rate 2016000
!
interface Serial0/0/3
 no ip address
 shutdown
 clock rate 2016000
!
ip default-gateway 192.168.1.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
ip http server
no ip http secure-server
!
!
!
!
!
control-plane
!
!
!
!
!
!
!
dial-peer voice 1 voip
 session target ipv4:192.168.1.4
 incoming called-number 9.......
 dtmf-relay h245-alphanumeric
 codec g711alaw
!
dial-peer voice 2 voip
 destination-pattern 9........
 session protocol sipv2
 session target dns:sip.simtex.com.au
 dtmf-relay rtp-nte
 codec g711alaw
!
sip-ua
 authentication username {user} password {password} realm sip.simtex.com.au
 registrar dns:sip.simtex.com.au expires 3600
!
!
!
!
line con 0
 logging synchronous
line aux 0
line vty 0 4
 login local
!
scheduler allocate 20000 1000
!
end

-----------------------------------------------------------------------------------------------------

Notes:

Call manager is at 192.168.1.4
Gateway 192.168.1.1
Cube is at 192.168.1.16

SIP server provider  domain is sip.simtex.com.au

[Aaron Harrison]

Hi 

debug ccsip messages

... will be much more helpful :-)

Aaron

[Areyouserious]

Will do.

[Areyouserious]

Had some sip commands and config examples sent to me by my provider.

I tried entering the commands they gave me,  some of them didnt work for some reason.

Tried upgrading the ios,  that didnt help either.

Ive included the new running config and the debug below.

Building configuration...

Current configuration : 2079 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Telephones
!
boot-start-marker
boot-end-marker
!
!
no aaa new-model
no network-clock-participate wic 0
!
!
ip cef
!
!
ip name-server 192.168.1.1
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
!
!
voice-card 0
 no dspfarm
!
!
!
voice service voip
 supplementary-service h450.12
 fax protocol none
 sip
  registrar server expires max 3600 min 3600
!
!
!
!
!
!
!
!
!
voice translation-rule 1111
 rule 1 /^...$/ /618********/
!
!
voice translation-profile PSTN_Outgoing
 translate calling 1111
!
!
!
!
username admin privilege 15 password 0 G54HW89Y4G
!
!
!
!
!
!
!
interface FastEthernet0/0
 ip address 192.168.1.16 255.255.255.0
 duplex auto
 speed auto
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/0/0
 no ip address
 shutdown
 clock rate 2016000
!
interface Serial0/0/1
 no ip address
 shutdown
 clock rate 2016000
!
interface Serial0/0/2
 no ip address
 shutdown
 clock rate 2016000
!
interface Serial0/0/3
 no ip address
 shutdown
 clock rate 2016000
!
ip default-gateway 192.168.1.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
!
ip http server
no ip http secure-server
!
!
!
!
control-plane
!
!
!
!
mgcp behavior g729-variants static-pt
!
!
!
dial-peer voice 1000 voip
 description AU 8-digit dialling
 translation-profile outgoing PSTN_Outgoing
 preference 1
 destination-pattern 0[2-9]........
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
sip-ua
 credentials username {SIPUSER} password 06525C741B realm sip.simtex.com.au
 authentication username {SIPUSER} password 135144475C
 no remote-party-id
 retry invite 2
 retry register 10
 timers connect 100
 registrar dns:sip.simtex.com.au expires 3600
 sip-server dns:sip.simtex.com.au
 connection-reuse
 g729-annexb override
!
!
!
!
line con 0
line aux 0
line vty 0 4
 login local
!
scheduler allocate 20000 1000
!
end

-------------------------------------------------------------

Debug CCSIP

-------------------------------------------------------------

hones#
Telephones#
*Feb 13 11:51:11.499: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Feb 13 11:51:11.499: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x458BB870, addr=192.168.1.4, port=59660, connid=1, transport=tcp
*Feb 13 11:51:11.499: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:992211882@192.168.1.16:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.4:5060;branch=z9hG4bK24a5f2178
From: <sip:2001@192.168.1.4>;tag=38~b177dfaa-b695-4ac5-a0ff-e2211bef2801-24342223
To: <sip:992211882@192.168.1.16>
Date: Fri, 13 Feb 2015 11:09:37 GMT
Call-ID: cae33400-4dd1db71-25-401a8c0@192.168.1.4
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Call-Info: <sip:192.168.1.4:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3403887616-0000065536-0000000037-0067217600
Session-Expires:  1800
P-Asserted-Identity: <sip:2001@192.168.1.4>
Remote-Party-ID: <sip:2001@192.168.1.4>;party=calling;screen=yes;privacy=off
Contact: <sip:2001@192.168.1.4:5060;transport=tcp>
Max-Forwards: 70
Content-Length: 0

*Feb 13 11:51:11.503: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.1.4,Port 59660, Transport 2, SentBy Port 5060
*Feb 13 11:51:11.503: //-1/CAE334000000/SIP/State/sipSPIChangeState: 0x460D2E80 : State change from (STATE_NONE, SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
*Feb 13 11:51:11.503: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.1.4,Port 59660, Transport 2, SentBy Port 5060
*Feb 13 11:51:11.503: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Feb 13 11:51:11.503: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.1.4,Port 59660, Transport 2, SentBy Port 5060
*Feb 13 11:51:11.503: //-1/CAE334000000/SIP/Info/sipSPIProcessCallInfoHeader: Callinfo parsed - host = 192.168.1.4, port = 5060, duration = 500

*Feb 13 11:51:11.503: //-1/CAE334000000/SIP/Info/sipSPIUASSessionTimer: Min-SE Header: 1800
*Feb 13 11:51:11.503: //-1/CAE334000000/SIP/Info/sipSPIUASSessionTimer: SE Value: 1800
*Feb 13 11:51:11.503: //-1/CAE334000000/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
*Feb 13 11:51:11.503: //-1/CAE334000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x460D2E80 key=cae33400-4dd1db71-25-401a8c0@192.168.1.4992211882
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 9922118820
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 2001
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name , number 2001, Calling oct3 0x00, oct_3a 0x80, Called number 992211882
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 3
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIContinueNewMsgInvite: Calling name , number 2001, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number 992211882, oct3 0x00
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
*Feb 13 11:51:11.507: //-1/CAE334000000/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
*Feb 13 11:51:11.507: //2/CAE334000000/SIP/Info/sipSPIDoQoSNegotiation: No SDP body
*Feb 13 11:51:11.507: //2/CAE334000000/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
*Feb 13 11:51:11.507: //2/CAE334000000/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 2, sdp 0x45F42C8C channels 0x460D4A30
*Feb 13 11:51:11.507: //2/CAE334000000/SIP/Info/copy_channels:
 callId 2 size 0 ptr 0x4515ED58)
*Feb 13 11:51:11.507: //2/CAE334000000/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Delayed media case...creating new stream
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 192.168.1.16
*Feb 13 11:51:11.511: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 19146 for stream 1
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=19146
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = cae33400-4dd1db71-25-401a8c0@192.168.1.4
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
*Feb 13 11:51:11.511: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateRawMsg: No GTD passed.
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS_DELAYED_MEDIA
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x460D2E80 key=cae33400-4dd1db71-25-401a8c0@192.168.1.44B620-44E
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 2 to table
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Transport/sipSPITransportSendMessage: msg=0x462EBBD0, addr=192.168.1.4, port=59660, sentBy_port=5060, is_req=0, transport=2, switch=0, callBack=0x00000000
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Feb 13 11:51:11.511: //2/CAE334000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Feb 13 11:51:11.515: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x462EBBD0, addr=192.168.1.4, port=59660, connId=1 for TCP
*Feb 13 11:51:11.515: //2/CAE334000000/SIP/State/sipSPIChangeState: 0x460D2E80 : State change from (STATE_IDLE, SUBSTATE_NONE)  to (STATE_RECD_INVITE, SUBSTATE_NONE)
*Feb 13 11:51:11.515: //2/CAE334000000/SIP/Info/sipSPIProcessContactInfo: Previous Hop 192.168.1.4:5060
*Feb 13 11:51:11.515: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
*Feb 13 11:51:11.519: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/Info/act_recdinvite_disconnect: Performing disconnect
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/Info/sipSPIValidateGtd: No rawMsg from CCAPI
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/Transport/sipSPITransportSendMessage: msg=0x462EBC60, addr=192.168.1.4, port=59660, sentBy_port=5060, is_req=0, transport=2, switch=0, callBack=0x41091834
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
*Feb 13 11:51:11.519: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x462EBC60, addr=192.168.1.4, port=59660, connId=1 for TCP
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/Info/sentErrResDisconnecting: Sent an 3456XX Error Response
*Feb 13 11:51:11.519: //2/CAE334000000/SIP/State/sipSPIChangeState: 0x460D2E80 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Feb 13 11:51:11.519: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.4:5060;branch=z9hG4bK24a5f2178
From: <sip:2001@192.168.1.4>;tag=38~b177dfaa-b695-4ac5-a0ff-e2211bef2801-24342223
To: <sip:992211882@192.168.1.16>;tag=4B620-44E
Date: Fri, 13 Feb 2015 11:51:11 GMT
Call-ID: cae33400-4dd1db71-25-401a8c0@192.168.1.4
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Content-Length: 0

*Feb 13 11:51:11.523: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 192.168.1.4:5060;branch=z9hG4bK24a5f2178
From: <sip:2001@192.168.1.4>;tag=38~b177dfaa-b695-4ac5-a0ff-e2211bef2801-24342223
To: <sip:992211882@192.168.1.16>;tag=4B620-44E
Date: Fri, 13 Feb 2015 11:51:11 GMT
Call-ID: cae33400-4dd1db71-25-401a8c0@192.168.1.4
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=1
Content-Length: 0

*Feb 13 11:51:11.527: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
*Feb 13 11:51:11.527: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessNewConnMsg: gConnTab=0x458BB870, addr=192.168.1.4, port=59660, connid=1, transport=tcp
*Feb 13 11:51:11.527: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:992211882@192.168.1.16:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.4:5060;branch=z9hG4bK24a5f2178
From: <sip:2001@192.168.1.4>;tag=38~b177dfaa-b695-4ac5-a0ff-e2211bef2801-24342223
To: <sip:992211882@192.168.1.16>;tag=4B620-44E
Date: Fri, 13 Feb 2015 11:09:37 GMT
Call-ID: cae33400-4dd1db71-25-401a8c0@192.168.1.4
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

*Feb 13 11:51:11.527: //2/CAE334000000/SIP/Info/sipSPIFindCcbUASReqTable: *****CCB found in UAS Request table. ccb=0x460D2E80
*Feb 13 11:51:11.531: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.1.4,Port 59660, Transport 2, SentBy Port 5060
*Feb 13 11:51:11.531: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Clock Time Zone is UTC, same as GMT: Using GMT
*Feb 13 11:51:11.531: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.1.4,Port 59660, Transport 2, SentBy Port 5060
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIIcpifUpdate: CallState: 2 Playout: 0 DiscTime:30879 ConnTime 0
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/State/sipSPIChangeState: 0x460D2E80 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE)  to (STATE_DEAD, SUBSTATE_NONE)
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x460D2E80
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 2001
Called Number            : 992211882
Source IP Address (Sig  ): 192.168.1.16
Destn SIP Req Addr:Port  : 192.168.1.4:5060
Destn SIP Resp Addr:Port : 192.168.1.4:59660
Destination Name         : 192.168.1.4

*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0
Source IP Address (Media): 192.168.1.16
Source IP Port    (Media): 19146
Destn  IP Address (Media): 0.0.0.0
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): 0.0.0.0:0

*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 1
Disconnect Cause (SIP)   : 404

*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIUdeleteccCallIdFromTable: Removing call id 2
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x460D2E80 key=cae33400-4dd1db71-25-401a8c0@192.168.1.4992211882
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x460D2E80 key=cae33400-4dd1db71-25-401a8c0@192.168.1.44B620-44E
*Feb 13 11:51:11.531: //2/CAE334000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
*Feb 13 11:51:11.535: //2/CAE334000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 460D2E80

[Vivek Batra]

Aaron will better comment if everything is fine between CUBE and service provider but in the above configuration, there is no dial-peer attaching to incoming call leg and destination-pattern 0[2-9]........ is matching the actual called number (994459020).

[Areyouserious]

hmmm, there was bound to be something wrong....

Thanks Vivek....I'll have a look into it.

Might also mention that some of the commands my sip provider gave me didn't work.

These were the commands:

• no supplementary-service sip moved-temporarily
• no supplementary-service sip refer
• localhost dns:sip.simtex.com.au
• no update-callerid
• voice-class codec 1 
• voice-class sip dtmf-relay force rtp-nte
• keepalive target dns:sip.simtex.com.au
• timers keepalive active 100
• host-registrar

[Aaron Harrison]

Hi

Firstly, Vivek is correct (+5). Your dial-peer has a destination-pattern of 0[2-9]........, and you are sending 9........ - there is no leading 0. You may be stripping this on CUCM (predot discard or similar). 

I'm not sure what your calling number is meant to be - is 994459020 valid? Should it have the 9s at the start? 
 

You can see the 'received' INVITE from CUCM, followed by a '404 not found' sent back as it doesn't match any peer.

Aaron

[Areyouserious]

Yes Our local household outbound phone numbers over here in West Australia are an 8 digit number usually starting with a '9',   and the extra '9' in there is currently the number i'm using to dial to an outside line from the Cisco phones, and yes that  extra 9 was specified in CUCM.

I did notice that 404 and figured it might by a point to have a look at.

Thanks Aaron,  and Thanks Vivek.

Ill have another look tomorrow morning into those to points you both kindly pointed out to me. (its 11:27pm over here right now, getting a bit late).

Thanks again.