CUCM 7 - Route incoming call to a specific VM

Hello Forum,
Hope this is an easy one for you guys.
I am trying to accomplish the following and not sure if its possible.
Ex.
Incoming call to ext. 1111
Ext. 1111 gets added to 2 phones, then call rings simultaneously on said 2 phones
After 5 rings and no answer, it rolls over to a different already existing VM (ext 8888)
Any help would be appreciated as I am not sure where to change the settings to make this happen.
Thanks in advance,
LS

Assuming x1111 does not need a dedicated voicemail, you could do the following:
Ensure that x1111 has the correct VM profile and set the CFB/CFNA settings to forward to VM as usual. On the mailbox for 8888, set an alternate extension of 1111.
Hailey
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Message was edited by: deggie

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Hi,
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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

Sir recd your reply for FaceTime my router is perfect at the.same time my I phone is receving incoming call

Sir recd your reply my router is perfect at the same time my I phone is receving incoming call

Is there a question here?

Iphone 4S; Phone & audio piece; Incoming calls are NOT routed automatically??

When I dial out; no problem the calls automatically goes through ear pice.
When I get incomming calls the call goes to phone, not even to the speaker in in the phone. So I have to
1.Slide the bar (then first realizing (again) dam: I hear nothing)
2. Choose the box for speaker options
3. Choose my blue ant piece..
By then some callers think I am not picking up or there is somthing wrong..!
My 3 colleaguers have same phone and no such problems. We are at a loss and I could not find reply in community groups. There was note about Iphone 5 and not correcting to Car speaker automatically. This was noted as a softwre issue Apple might address in next version. But since my colleagues do not have problem with their Iphone it must be either my settings or a weird fault with my phone.
One of my collegues have same ear pice as me, so the issueis not that either.
Is there some weird setting I need to fix to correct the problem?
Thank you

Are you able to make calls? I would try a couple different things -
1. Try doing a double hard reset - Press and hold home button and the sleep wake button at the same time until the screen displays the slide to shut down red button. Continue pressing and holding both the buttons until the screen goes black and then the apple logo flashes up. Continue pressing and holding both the home and sleep wake button beyond this point until the screen goes black again. At this point, let go of the home button and press once the sleep wake button as you would to start your phone. See if this double hard reset helps clear out any cache.
2. If this fails, I would try restoring the phone as a new phone in iTunes. Backup your phone before this so that you dont lose your contacts, etc. You can do a couple test calls once the firmware is installed before setting up the phone, data, apps, etc. If it looks good, try restoring your phone from an earlier backup.
If everything fails, call AppleCare or see a Genius in the store.

SIP ITSP on CUCM 10.5.2 (No CUBE) Incoming calls fail, outgoing are fine

Hi,
I am in the process of upgrading a customer who is on 8.0.3. They have an ITSP terminating SIP Trunk directly on the CCM Server
I upgraded the system to 10.5.2. During cutover I was able to make outgoing calls but all incoming calls were failing.
After reverting back to the old system, everything is working fine again, and I dont understand what could be the possible issue that it doesnt work on 10.5.2 but it works well on 8.0.3.
I checked almost everything and dont find anything that stands out, which may be contributing to the issue.
Any idea what could be missing here?
Thanks

Thanks for all your tips.
It was turned out that, the URI was a FQDN and during the first install of the 8.0.3 (in the sandbox) I had not bothered to get the DNS Services replicated and then didnt check if the ITSP was sending the invite on URI based on FQDN or IP Address
Thanks

Matching B-channel for dialpeer assigment (Incoming call)

Is it possible to manipulate a specific B Channel to use a dialpeer? For example on a T1 with 24 channels, I need the last 4 B-channels to choose an specific dialpeer, same concepts as matching incoming DID's but on a B-Channel slot.
If a call arrives on B-channel 20, I need that call to be sent to a specific DN at the CallManager.
I am using H.323 on the gateway side.
thanks in advanced.
Oscar

Hi!
I'm stuck in a similar situation at the moment, and hope someone has solved this issue.
I have a "back to back connection" between two PBX'es that communicate using Q.931. I have had to replace my old hardware running a legacy CCS solution due to its incapability of understanding overlap signalling. I have replaced it with two Cisco routers running E1 (Q931). Between the units I have a high latency low bandwidth network.
The issue is that the PBX'es are configured in a way that requires calls between the PBX'es to use the same time slot at both ends. I'm running a VoIP network with CUCM between the two sites.
=============================================
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-21
trunk-group TS01 timeslots 1
trunk-group TS02 timeslots 2
trunk-group TS03 timeslots 3
trunk-group TS04 timeslots 4
trunk-group TS05 timeslots 5
trunk-group TS06 timeslots 6
trunk-group TS07 timeslots 7
trunk-group TS08 timeslots 8
trunk-group TS09 timeslots 9
trunk-group TS10 timeslots 10
trunk-group TS11 timeslots 11
trunk-group TS12 timeslots 12
trunk-group TS13 timeslots 13
trunk-group TS14 timeslots 14
trunk-group TS15 timeslots 15
trunk-group TS17 timeslots 17
trunk-group TS18 timeslots 18
trunk-group TS19 timeslots 19
trunk-group TS20 timeslots 20
trunk-group TS21 timeslots 21
dial-peer voice 81030001 pots
trunkgroup TS01
description ** PBX TS01 **
translation-profile incoming 61031401
translation-profile outgoing 25
destination-pattern 81030001T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
dial-peer voice 81030002 pots
trunkgroup TS02
description ** PBX TS02 **
translation-profile incoming 61031402
translation-profile outgoing 25
destination-pattern 81030002T
progress_ind alert enable 8
progress_ind progress enable 2
incoming called-number .
no digit-strip
voice translation-rule 25
rule 1 /^810300../ //
voice translation-rule 61031401
rule 1 // /61031401\1/
voice translation-rule 61031402
rule 1 // /61031402\1/
voice translation-profile 25
translate called 25
voice translation-profile 61031401
translate called 61031401
voice translation-profile 61031402
translate called 61031402
=====================================
The idea is that I do not know (or care) what numbers are used as SOURCE or DESTINATION of the original call. My network should be transparent to the PBX number plan. I need to add a prefix, and it should be based on the timeslot the call comes in on. I route the traffic between the routers using the prefix.
The configuration excerpt above should add 61031401 prefix to all calls entering on TS01, and 61031402 to all calls entering on TS02 etc. Calls from the remote should have corresponding prefixes 81030001 for TS01 and 81030002 for TS02 etc.
The outbound (from voip to pots) routing of the above configuration works.
However I have a challenge with the incoming prefixing.
All calls inbound end up using "dial-peer 81030001 pots".
I believe the reason this dial-peer "takes" all of the calls inbound from pots is due to the line "incoming called-number ."
Removing this makes no inbound pots call work as the "destination-pattern 8103001T" is never matched.
Removing "destination-pattern 8103001T" from the dial-peer is not working as it kills the voip to pots routing of inbound calls from the remote router.
Anyone got a good idea for me?

CUCM 7 - Route incoming call to a specific VM

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