CUCM Wake-Up Call

I need a product or feature to schedule a call to an IP Phone for use as a Wake-Up call for hotel guests.
This can either be scheduled by the guest themselves or by reception.
IPCelerate sounds like it could offer this, but the price says no ($31k list).
Has anyone else come across anything similar?

Do you still have this? I am new to CUCM and trying to figure out how to do this. Visited your site, but nothing seems to be free... Would you be able to provide a sample of how this is done or a setup guide on how to implement what you have created? Thanks in advance

Similar Messages

Wake up calls

I have a new landline number and for some reason there is a wake up call set for 5am. I have already called once and the agent could not find it on the account. i had just set up the voice mail. The agent said that should clear out any previous setting. I am still getting these wake up calls and I dont know how to take them off. There is no menu option to delete a wake up call. There is only options to set up wake up calls.. Please help...

There is nothing really built in these apps to do that easily, but there are plenty of 3rd party solution that integrate with CUCM for hospitality market. Novotek comes to mind.
Chris

Wake Up Call to Extension

Hey Guys,
Does anybody know of a free service that allows you to send wake up calls to an extension through specified phone number? It would really help me out a lot.
Thanks!

You can use my script "alarm call". for CUCME and CME. It can be acquired on the website mentioned in my profile.

CUCM 8.6 Call Forwarding to External Number Issue

Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
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
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
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
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
   Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
   Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source           IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone

How to make "efficient" wake-up call?

Hi,
Are there any means to force I-Phone to ring longer than 2 seconds for wake-up call? I can not find any setting to make wake-up call ezcepted "meeting" or so in the calendar. Is ther something you can suggest?
Thanks

Is it possible to make an alarm for an event in the calendar app continue to ring until you shut it off?

TS3938 The message "PowerPC applications are no longer supported" sounds like the end, not an interruption. Nevertheless the gaming world could use a wake up call. Is this really a solution?

The message "PowerPC applications are no longer supported" sounds like the end, not an interruption. Nevertheless the gaming world could use a wake up call. Is this really a solution?

Scot Lee wrote:
The message "PowerPC applications are no longer supported" sounds like the end, not an interruption. Nevertheless the gaming world could use a wake up call. Is this really a solution?
Yes Rosetta has been gone for over two years. You can always search here for how Michael Lax's instructions on how to run SL Server in a virtul environment, but I don't think it is too kind to gamers.
Advice? Buy an old Mac that runs PPC software.
Cheers
Pete

ROBO call blocker is blocking my wake up calls

I've had wake up calls for 3 years using my FIOS landline voice mail. I also Implemented robocall blocker about a year ago.
Within the past week, the blocker is now stopping the wake up calls.
Verizon, how do I get around this? I've added the voicemail number that calls with the wake up call to my accepted list,
still no resolution.

dezyndiva wrote:
that's part of the problem. I don't recall now how I set it up. However, It's illogical that I would have to turn it off
just to get wake up calls. Why now is the blocker recognizing the wake up call as coming from a robodialer,
it never blocked it for the past year.
With FiOS digital voice, there is no native "ROBO call blocker" so you must be using something else. If you're using something like nomorobo then it uses the simulatenous ring feature which can be disabled if nomorobo (or some other service like it) is blocking the call. You check your simulataneous ring setup.

CUCM 8.6 call busy between SIP phones and thirdparty phones

Hi Everybody,
I have the following error on my logs:
Invalid Disconnect Cause(cause=47), No Reason Header Appended
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getXCiscoViPRFallbackIDAndDTMFKey: Device type 8, Pstn Fallback is not enabled|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefCcRegister: Secure status=1, mSrtpPresent=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/compareAndUpdateMedia: sdpStatus=0, CMEndPointSDP role=1, SIPEndPointSdpRole=2|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefCcRegister: Secure status=1, mSrtpPresent=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPUACSessionExpires: isMidCall[0], response[200], method[102]|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/parseSessionExpires: refresh_interval[1800], refresher[uas]|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/setSIPSessionExpiresTimer: interval[1768] secs|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSecureRec: enforce srtp flag: 0, remote end srtp support: 0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/updateCNToCC: identityCngFlag[0x1f], isConnInfoInd[1], ccContactHeader[]|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPConnectInd: Exit with state = outCall_200Rcvd|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/handleSIPConnectInd: Exit with state = outCall_200Rcvd|3,100,63,1.36454459^192.168.0.15^*
14:46:50.091 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/setSIPUpdateFlags: mIsUpdateForSignalingAllowed = 1 mIsUpdateForMediaAllowed = 1 mPendingOutgoingUpdate = 0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/addTransparencyInfo: attaching transparency object|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPCdpc(3,74,281707)/ci=144600614/ccbId=35257792/scbId=0/getDefAe: SIPCdpc=281707, nodeId=3, processNumber=73 ci=144600614, branch=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |MRM::waiting_MrmDeallocateMtpResourceReq- Deallocate received for CI=53993831 count=0|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |MRM::waiting_DeallocateMtpResourceReq- ERROR Deallocate received for an unknown Call Identifier Ci = 53993831|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_DISCONNECT value=500 retries=10|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_DISCONNECT value=500 retries=10|3,100,63,1.36454459^192.168.0.15^*
14:46:50.098 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 192.168.0.15:[5060]:
the calling number is 34967850938
the callied number is 19026

Julien,
Please use the link be low to collect cucm traces and use the advanced editor on the forum (located on top right hand corner of the discussion widnow) to attach the trace
https://supportforums.cisco.com/docs/DOC-29901
Ensure you collect the trace from the folowing
1. the server that the phone is registered to
2. If this server is different from the server in the cucm group of the sip trunk, then you need to also collect traces from the server (s) in the cucm group assiged to the sip trunk that connects to the 3rd party cluster...
NB: If you have three servers in the cucm group of the sip trunk, you have to collect the trace from all three servers. This is because calls are dsitributed in a round robin fashion to servers in a sip trunk...
FInally before you send the trace over, please ensure the calling and called numbers are present. Also include the time of the test call
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Avaya calls to CUCM 8.6: Call Park/Hold Fail

This may be similar to the issue I posted here: https://supportforums.cisco.com/message/3496110#3496110
4 digit dialing from an Avaya system to a CUCM 8.6.2 server works fine, the Avaya is set as an h.323 gateway in CUCM.
Once the call is active if the Cisco phone places the Avaya caller on hold they hear no hold music and the call stays active until the Cisco phone tries to retrieve it. Then it's a fast busy on both end.
The same thing happens with calls placed on park.
Any suggestions?

Hi
Are you using G711 or G729 for these calls? If using G729 the CUCM software MTP cannot be used as it only supports g711, you would have to use a gateway-based MTP.
Try setting the calls to G711 to see if it works as a test.
Aaron

CUCME 8.6 Call not forwarding Voicemail

Hi frieds,
     In our office we are using CUCME 8.6 on Cisco 2951 and unity express 8.5 in ISM module. As per our configuration whenever user is busy or not answering , the call will forward to voicemail. Totally we have 24 PSTN line. So we have an additional gateway 2901. The Issue I’m facing is that, when a PSTN incoming call coming through the second gateway(2901), if the extension is busy or not answering the call is disconnecting instead of forwarding to voicemail.
My 2951 configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server.
Dial peer we are using for voice mail:
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad.
2901 Configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
ipv4 172.16.19.82
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad
============================
Debug CCSIP Calls
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0xAF40FD8
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 5000
Called Number            : 1099
Source IP Address (Sig ): 172.16.19.80
Destn SIP Req Addr:Port :
Destn SIP Resp Addr:Port :
Destination Name         :
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 172.16.19.80
Source IP Port    (Media): 25364
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 47
Disconnect Cause (SIP)   : 200
For your reference I here attach a network diagram
What the command which I missed?

Check License status on your CUE, I had same issue.. Finally figured out its about license.. sh license status
Sent from Cisco Technical Support iPhone App

CUCM to CVP calls. CTI-RP vs Route Pattern

CVP 9 or above
CUCM 9 or above
Requirement:
1. Consultive Warm Transfer - The agents to be able to transfer calls to a a different department by dialing an internal number and wait in the queue until answered.
2. Internal - Back-office people to dial internal IT-Helpdesk or HR
I see the above call flows as same, i.e. a Call Originating from CUCM to CVP .... correct me please?
I have tested both and they both work exactly the same way, i.e. using a CTI-RP associated to PGUSER, ICM answers it sends correlation id to CUCM and CUCM sends this to CVP ...AND... using a simple route patters instead point to CUCM-CVP SIP trunk. Functionally they behave same way - ICM/CVP answers and queues call until answered.
But the documentation confuses me, below snippet from CVP Config Guide
"... Calls Originated by Unified CM
Internal Help Desk calls: For these calls, the Unified Communication Manager (CM) phone user calls a CTI Route Point
Consultative Warm Transfer: For these calls, a Unified CM agent places the caller on hold and dials in to Unified ICME to reach a second agent .... "
And then on the same doc, there a Note
(*1) Note For warm transfers, the call from Agent 1 to Agent 2 does not typically use a SIP Trunk, but you must configure the CTI Route Point for that dialed number on the Unified CM server and associate that number with your peripheral gateway user (PGUSER)
(*2) And then again on the same doc under 'Unified ICME Warm Consult Transfer/Conference to Unified CVP' chapter/section it mentiones doing this using a Route Patter 'Create a route pattern and assign the route list to the route pattern'
So the confusion is
1. Why treat these call flows as Internal and Warm Transfer - they are calls from CUCM to CVP for the same end result - queue the call and transfer to an agent?
2. Route pattern or CTI-RP, what diff it makes? They both behave the same way, so is there a diff from reporting point of view that a call to CTI-RP are treated as Transferred rather than new calls or what?
3. Also if you compare (*1) & (*2) above, they both talk about Warm Transfer and *1 says 'must use CTI-RP' and *2 says use a Route Pattern?
Please assist.
Thanks & Regards,
Kartik

Kartik,
The Route Pattern that is mentioned is used for connecting a call leg through CVP to a local VXML Gateway for media playback. The CTI Route Point is entirely different from the Route Pattern/Route List setup. Here's the basic call flow:
Internal caller dials DN
DN hits CTI RP in CUCM. CTI RP sends call to ICM.
ICM matches DN to Call Type to Script, executes Script.
At some point, Script has either Send To VRU or a Run Ext. Script node.
ICM sends CUCM Network VRU label back to CUCM.
CUCM routes label using Route Pattern and Route List. The CSS of the internal caller determines how this is modified, i.e. which prefix digits to add for determining VXML gateway to route to.
Call is sent to CVP through SIP trunk
CVP receives call, tells ICM it has the call.
CVP starts new call leg to VXML gateway with digit string to match bootstrap dial-peer.
VXML Gateway receives call, initiates bootstrap TCL and VXML magic.
Yes, this is basically the same call flow for a fresh internal call to a queue, or an internal warm transfer to a queue. The CTI Route Point is needed in both cases. The Route Pattern/Route List combo is needed in both cases.
When you start looking at reporting, yes of course the two call scenarios are different. One is a transfer, the other isn't. The transferred call will have a more complex call history if you look at it in the TCDR.
From the standpoint of call legs, you will use less legs if you do a direct (one-step) transfer instead of a warm transfer. It is also simpler to maintain the call context in that case. In a warm transfer scenario, the agent is putting a caller on hold, then starting a new call, and joining the two calls together. The new call is coming from the agent, not the original caller. In a direct transfer, CVP just takes back the original call, potentially does more queuing, then sends the original caller to a new agent target.
-Jameson

How to get logs from CUCM for initiated calls

Dears,
like in Voice Gateway for initiated calls we can run debugs command like debug voip ccapi inout to know the calling and the called number and the status of the call, matched dial-peer, ... etc
Can we do the same in CUCM
Thanks in advance for your help

CDR is used mainly for call statistics, it is not designed for detailed troubleshooting. Reading cucm logs requires a detailed understanding of how different components function and some practice. It can be a little daunting.
This book is excellent. This is what I always recommend as the starting point..
http://books.google.co.in/books?id=YcCjKsssklIC&pg=PA42&lpg=PA42&dq=call+manager+trace+reading&source=bl&ots=ZW5_hvNFGA&sig=22rtALJZjQFfe9JrHjgAB_GgtRU&hl=en&ei=4qT3TLfTDsuYOs_I3YwI&sa=X&oi=book_result&ct=result&redir_esc=y#v=onepage&q=call%20manager%20trace%20reading&f=false
This doc explains how to read h323 traces in cucm
https://supportforums.cisco.com/docs/DOC-11779
This blog explains how to understand sip traces..
https://supportforums.cisco.com/document/113271/understanding-sip-traces

Use Cisco CUCM for outbound "call me at" feature on Lync meetings

I'm trying to find a step by step to enable users (non enterprise voice users) to use the dial me at feature in Lync conference meetings. I only want the user to have the ability to tell Lync to dial a number to place that number into the conference call,
the feature is easy to enable but i can't get the routing right between CUCM and Lync. I've looked all around the net but I can't seem to find anything that matches what i'm trying to do, other docs cover enterprise voice and that's out of my scope. Any assistance
here would be nice. Thanks

Hi,
In Lync Server 2010, it is not supported with “call me at” function for non-Enterprise Voice users.
However, Lync Server 2013 now allows participants that are not Enterprise Voice enabled to initiate dial-out calls from a meeting conference, called “Dial-out Conferencing for non-Enterprise Voice users”.
This can be configured by setting the Conferencing policy to allow this feature (Set-CSConferencingPolicy –AllowNonEnterpriseVoiceUsersToDialOut:$true). After enabling this, then assign a voice policy to the users who need the function.
Best Regards,
Eason Huang
Please remember to mark the replies as answers if they help, and unmark the answers if they provide no help. If you have feedback for TechNet Support, contact [email protected]
Eason Huang
TechNet Community Support

CUCME Not Incoming Calls, Outgoing calls ok

Hello everybody,
i am configuring a CUCME with SIP trunk, i can make calls to outside but i can´t recieve any from outside, this is my second time a configure with SIP
i´ve used the command debug voice dialpeer all to check was going on, but i can´t find the problem.
this is my config:
ip host sip-server A.B.C.D
voice service voip
ip address trusted list
ipv4 A.B.C.D 255.255.255.252
voice translation-rule 1
rule 1 /325277/ /1\1/
voice translation-profile IN
translate called 1
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming IN
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
ephone-dn 1
number 100
description RECEPTION
ephone 2
mac-address AAAA.BBBB.CCCC
ephone-template 1
type 7942
keep-conference
button 1:1
NOTE: IP Address are hidden, just for security
These are the output of my debug/tests:
#test voice translation-rule 1 32527700
Matched with rule 1
Original number: 32527700       Translated number: 100
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=32527700
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=32527700, Expanded String=32527700, Calling Number=32527700T
   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
   dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=NO_MATCH(-1)
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=59513212, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ANSWER; Calling Number=59513212
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=59513212T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_ORIGINATE; Calling Number=59513212
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=59513212T
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
*Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:exit@6704
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_VIA_URI; URI=sip:A.B.C.D:5060
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_REQUEST_URI; URI=sip:[email protected]:5060;user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_TO_URI; URI=sip:[email protected];user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_FROM_URI; URI=sip:[email protected];user=phone
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Dial String=, Expanded String=, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Result=-1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=32527700
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
   Is Incoming=TRUE, Number Expansion=FALSE
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
   Dial String=32527700, Expanded String=32527700, Calling Number=
   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:
   Result=Success(0); Incoming Dial-peer=1 Is Matched
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:exit@6076
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:
   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:exit@6655
Can Anyone help me???
Thanks in Advance!!!

Thanks, these are the output
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:32527700@(WAN):5060;user=phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
From: ;tag=6e8b9968-CC-25
To:
CSeq: 1 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Max-Forwards: 70
Supported: 100rel,timer
User-Agent: Huawei SoftX3000 V300R601
Session-Expires: 300
Min-SE: 90
Contact:
Content-Length: 376
Content-Type: application/sdp
v=0
o=HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)
s=Sip Call
c=IN IP4 (SIP_SERVER)
t=0 0
m=audio 11554 RTP/AVP 8 0 18 4 2 98 98 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-40/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:98 G726-24/8000
a=ptime:20
a=fmtp:18 annexb=no
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=32527700, Called Number=32527700, Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=32527700
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
   dialstring=32527700, saf_enabled=1, saf_dndb_lookup=1, dp_result=-1
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=NO_MATCH(-1)
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=59513212, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=-1
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
   Calling Number=59513212, Called Number=32527700, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
*Jan 29 16:53:19: //-1/FB88A7CE80F0/DPM/dpMatchSafModulePlugin:
   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
From: ;tag=6e8b9968-CC-25
To: ;tag=4CD1E84-2094
Date: Wed, 29 Jan 2014 22:53:19 GMT
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Min-SE: 1800
Server: Cisco-SIPGateway/IOS-15.2.4.M3
Content-Length: 0
*Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:32527700@(WAN):5060;user=phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER):5060;branch=z9hG4bK776928550196f0d843ca0b092
Call-ID: SBC9722bb53005161bf8cca630444260574@SoftX3000
From: ;tag=6e8b9968-CC-25
To: ;tag=4CD1E84-2094
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
*Jan 29 16:53:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:(SIP_SERVER):5060 SIP/2.0
Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
From: ;tag=4CD4D7C-1634
To:
Date: Wed, 29 Jan 2014 22:53:31 GMT
Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M3
Max-Forwards: 70
Timestamp: 1391036011
CSeq: 66 REGISTER
Contact:
Expires: 3600
Supported: path
Content-Length: 0
*Jan 29 16:53:31: //973/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP (WAN):5060;branch=z9hG4bK3B11F0F
Call-ID: 3017EE62-885411E3-80B4FEFC-CAA82B4A
From: ;tag=4CD4D7C-1634
To: ;tag=f2056e8e
CSeq: 66 REGISTER
Content-Length: 0
I´ve replaced the IP Adress for (SIP_SERVER) / (WAN) / SIP_SERVER_INTERNAL
Thank you

CUCM 7 Blackberry call forwarding

Hello,
I have CUCM 7. I wish to call forward to a blackberry device. When this is attempted the ISDN origin and destination cause codes are 4. This states a special information tone is to be generated. Does anyone know a solution for this ?
Many thanks
Stephen
Cause No. 4 - send special information tone [Q.850]
This cause indicates that the called party cannot be reached for reasons that are of a long
term nature and that the special information tone should be returned to the calling party.

Hi, thanks for your response.
We need a SIP route pattern because the 3rd party server at the end of the SIP trunk uses SIP URI dialling when it talks back to the CallManager.
The requirements are that the call must first go over the SIP trunk to the 3rd party SIP server. If there is no response, then an alternate destination must be tried by the CallManager. This can be another SIP trunk to a second SIP server or it can be an DN internal to the CallManager.
Route groups can't be used because if a SIP trunk in configured and then assigned to a route group, the same SIP trunk can not then be referenced in a SIP route pattern.
In other words for communication to work with the 3rd party SIP server,
Step 1: The SIP trunk must be referenced directly by a route pattern.
Step 2: The SIP trunk must be referenced by a SIP route pattern.
We won't need route groups if the second destination is internal to the CallManager.
However, how do we hunt with 1 - Route pattern to SIP Trunk and then 2 - Internal DN?

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