Integrating CCM 4.1(3) w/ a SIP IVR

I have set up a SIP trunk to move calls between SCCP endpoints/MGCP gateways and a SIP IVR. I have assigned it to a Route Group/Route List and assigned a Route Pattern. The IVR is reachable, but the SIP trunk is always sending DNIS as the route pattern assigned to the SIP trunk (8000), rather than the original called number (7000) (the CCM-side device dials a number and is redirected across the SIP trunk). Is there a way to send original dialed number rather then the redirected number here?
Also, the SIP trunk's destination is set as the IP Address of the IVR, and not a proxy server. If this IVR is the only SIP device I need to support, will I have any trouble with this configuration?

David
I'm not sure I fully understand your configuration. I have a SIP trunk connected to a SIP Server providing call center duties.
I defined a SIP Trunk with the server name = Mesa. I have the fields Connected Line ID and Name = Allowed, Redirecting Number Delivery-Inbound/Outbound checked. I assigned a DN 0f 6000 to the SIP server. Then I built a route pattern 6000 with the Gateway or Route List field = Mesa. All of the fields on the route pattern are set to default. When someone on the CCM side dials 6000 the call is routed across the SIP trunk to the SIP server and the Caller ID is displayed correctly. When one of the SIP call agents makes an oubound call it routes across the SIP trunk and the Caller ID displays correctly. Hope that provides some help.
Larry

Similar Messages

Cisco SIP Phone 9971 won't register on CME 8.6

Hello,
I'm facing a very strange problem:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the related-postings to this and other Forum, but I have not been able to solve it.
One of the "potential solutions" was to make sure that the Phone had a Line configured.
But I think that the commands voice register dn and voice register pool are properly configured (see config below)
So frankly, I have no idea what I could be missing.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#

Thank you for your reply.
I did some debugs and the results are very strange!
This is what I got:
Feb 24 18:01:12.219: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK08011844
From: ;tag=189c5db6bd09000260cf3daf-289a76d1
To: ;tag=52488-160A
Date: Mon, 24 Feb 2014 18:01:12 GMT
Call-ID: [email protected]
CSeq: 1000 REFER
Content-Length: 0
Contact:
Feb 24 18:01:12.291: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:172.25.140.1 SIP/2.0
Via: SIP/2.0/UDP 172.35.140.12:5060;branch=z9hG4bK1e9ad079
From: ;tag=189c5db6bd0900032df02e9c-25d79707
To:
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 01 Jan 1982 00:02:41 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP9971/9.4.1
Contact: ;+sip.instance="
000000-0000-0000-0000-189c5db6bd09>";+u.sip!devicename.ccm.cisco.com="SEP189C5DB
6BD09";+u.sip!model.ccm.cisco.com="493";video
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-
cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-
cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-
8.0.1
Content-Length: 0
Reason: SIP;cause=200;text="cisco-alarm:22 Name=SEP189C5DB6BD09 ActiveLoad=sip99
71.9-4-1-9.loads InactiveLoad=sip9971.9-3-2SR1-1.loads Last=reset-reset"
Expires: 3600
Feb 24 18:01:12.395: voice_reg_get_reg_expires_timer: no voice register pool found
Feb 24 18:01:12.395: VOICE_REG_POOL: Register request for (1010) from (172.35.140.12)
Feb 24 18:01:12.395: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: key(1010) contact(172.35.140.12:5060) add to contact table
Feb 24 18:01:12.395: VOICE_REG_POOL: No entry for (1010) found in contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(1010) contact(172.35.140.12) added to contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) add to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: No entry for (172.35.140.12) found in srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL: key(172.35.140.12) contact(1010) added to srst contact table
Feb 24 18:01:12.399: VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
But right after these errors, I get the following:
Feb 24 18:01:12.399: VOICE_REG_POOL: Creating param container for dial-peer 4000
1.VOICE_REG_POOL pool->tag(1), dn->tag(1), submask(1)
VOICE_REG_POOL pool_tag(1), dn_tag(1)
Feb 24 18:01:12.399: VOICE_REG_POOL: Created dial-peer entry of type 0
Feb 24 18:01:12.399: VOICE_REG_POOL: Registration successful for 1010, registration id is 1
Feb 24 18:01:12.411: VOICE_REG_POOL: Contact matches pool 1 number list 1
Feb 24 18:01:12.411: VOICE_REG_POOL: GW SIS: X-cisco-cme-sis-1.0.0
Feb 24 18:01:12.411: VOICE REGISTER POOL-1 has registered.
                               Name:SEP189C5DB6BD09 IP:172.35.140.12 DeviceType:Phone
Feb 24 18:01:12.411: VOICE_REG_POOL: Pool[1]: service-control (reset type: 2) message sent to sip:[email protected]
Feb 24 18:01:12.411: voice_reg_privacy_update_to_phone: delay sending privacy update during bulk registration
Feb 24 18:01:12.415: //1/7B0070C28003/SIP/Msg/ccsipDisplayMsg:
====================
And when I do a sh voice register pool, I get the following:
C2811#sh voice register pool 1
Pool Tag 1
Config:
Mac address is 189C.5DB6.BD09
Type is 9971
Number list 1 : DN 1
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9971/9.4.1
DTMF Relay is enabled, rtp-nte
Call Waiting is enabled
DnD is disabled
Video is enabled
Camera is enabled
Busy trigger per button value is 0
call-forward b2bua busy 68600
keep-conference is enabled
registration expires timer max is 3600 and min is 120
username adm password adm
kpml signal is enabled
Lpcor Type is none
blf call list is enabled
Transport type is udp
service-control mechanism is supported
registration Call ID is [email protected]
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 1010
contact IP address: 172.35.140.12 port 5060
Phone SIS Version: 6.0.2
GW SIS Version: 1.0.0
Dialpeers created:
Dial-peers for Pool 1:
dial-peer voice 40001 voip
destination-pattern 1010
session target ipv4:172.35.140.12:5060
session protocol sipv2
dtmf-relay rtp-nte
digit collect kpml
codec g711ulaw bytes 160
no vad
call-fwd-busy        68600
after-hours-exempt   FALSE
Statistics:
Active registrations : 4
Total SIP phones registered: 1
Total Registration Statistics
    Registration requests : 4
    Registration success   : 4
    Registration failed    : 0
    unRegister requests    : 0
    unRegister success     : 0
    unRegister failed      : 0
    Attempts to register
           after last unregister : 0
    Last register request time   : 18:11:43.551 UTC Mon Feb 24 2014
    Last unregister request time :
    Register success time        : 18:11:43.551 UTC Mon Feb 24 2014
    Unregister success time      :
C2811#
So apparently the Phone is actually registered!
However, the Phone screens still shows this message: Phone Not Registered.
So frankly I don't understand what's going on!
I really hope somebody can help. Thanks!

CCM 2.0 with ECC 6.0

Hi Guys,
We are trying to integrate CCM 2.0 with ECC 6.0 on the purchasing application of ECC. The requirement suggests that the requester should be able to access the catalogue via OCI from ME51n (Create a PR) and select a good/service and add it to the Purchase Requisition directly from CCM.
I do not have experience with integrating CCM to ECC. I was hoping that someone in the community has worked on this before and can shed some light on how to achieve this requirement.
Suitable answers will be rewarded with points !!
Thanks,
Sundeep

Sundeep: both MM-to-CCM and CCM-to-MM integration is standard as of ECC 6.0.
Check the following documentation (talks more about MM-to-CCM, but there's some info on Catalog-to-MM):
<a href="http://help.sap.com/saphelp_erp2005/helpdata/en/46/b596e138a941ce9fba8fc8533674ee/frameset.htm">Integration of Web-Based Catalogs in Purchasing</a>
Cheers,
Serguei

NICE recorder - one way or garbled audio over SIP trunk.

Customer with CUCM 10.5.2 trying to integrate with a NICE recording solution. Everything configured per NICE documentation and rechecked that several times with the NICE vendor. However, if calling to/from the PSTN or legacy Nortel PBX to a Cisco 78xx or IP Communicator soft phone - we get one-way audio pushed out the SIP trunk to the NICE system. If it's a Cisco to Cisco phone call - the audio is garbled.
Has anyone experienced this issue with this type of integration - or the same issue with a SIP trunk to the CUCM to another system at all? We're at a loss here.
Thank you.

This document should help you:
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml

IPod as SIP client?

Hi,
is it possible to use the newest iPod touch as SIP client? So could the integrated microphone and speaker be used from SIP Apps?
Thank you.
Thorsten Dreiner

Did you look at the apps in the App store? The SIP by toofone app says it is also compatible with the iPod Touch with iOS 4 and later.

CallManager 7.0 Microsoft Lync 2010 Integration

We are currently running CUCM 7.0.2 and Microsoft Lync 2010 and want to be able to have Lync clients make calls from Lync. There are lots of documents that talk about the integration and this makes things very confusing (as to what to do). What are our options here? Which option is the best? Which options is the easiest?

Hi,
First of all you'll need to upgrade CUCM to 7.1.5 or later according to:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucimoc/8_5/english/release/cucimocReleaseNote.html#wp342335
CUCM 7.0 might work, but it was not tested. The integration needs to be done using a SIP trunk between the servers. You can refer to the guide below:
http://www.cisco.com/c/dam/en/us/solutions/collateral/enterprise/interoperability-portal/1048876.pdf
I also found this guide on the MS site:
http://www.microsoft.com/en-us/download/confirmation.aspx?id=26800
Hope this helps!
Regards,
Tere.

No SDP in INVITE on IP2IPGW

Hello,
I am testing a new version of IOS, Version 12.3(11)T2, on a 7204 VXR, for interworking H.323 and SIP. I've enabled the interworking in the IOS with the "allow-connections h323 to sip" command. I'm using a 7960 behind a Call Manager to initiate the H.323 calling. My test call-flow is as follows:
CCM 4.02-->H.323-->7204VXR-->--SIP-->NexTone
When I place a test call from the IP phone, the 7204VXR converts the H.323 call & sends out a SIP INVITE--but, without any SDP information. The missing SDP info that is NOT in the INVITE (endpoint IP address, preferred codec, etc) seems to be present in the upstream H.225 cs: setup messaging coming out of the Call Manager. Packet captures look like the CCM is using fast-start.
For some reason, if I use the "CSIM start xxxxxx" command in the IOS to generate a SIP test call directly from the 7204VXR to my NexTone, the SDP information is included in the INVITE.
Has anyone else had any luck with H.323/SIP interworking on the 72xxVXR? Is the omission of the SDP info in the SIP INVITES a bug, or a case of mis-configuration on my part?
Partial debug CCSIP message output is below.
Thank you,
SK
H.323-to-SIP has no SDP:
ENV1.LAB#term mon
ENV1.LAB#
*Feb 10 13:19:50.546: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.31.20.14:5060;branch=z9hG4bK7A28
From: <sip:[email protected]>;tag=23D59850-F33
To: <sip:[email protected]>
Date: Thu, 10 Feb 2005 13:19:50 GMT
Call-ID: [email protected]
Supported: 100rel,timer
Min-SE: 1800
Cisco-Guid: 2151945792-2627313949-654311424-1076826018
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
Timestamp: 1108041590
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.31.20.14:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
And now, the SDP info shows up with the "csim start command"
ENV1.LAB#csim start 7035551212
csim: called number = 7035551212, loop count = 1 ping count = 0
*Feb 10 13:21:05.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.31.20.14:5060;branch=z9hG4bK7D1C51
From: <sip:10.31.20.14>;tag=23D6BDFC-1BE7
To: <sip:[email protected]>
Date: Thu, 10 Feb 2005 13:21:05 GMT
Call-ID: [email protected]
Supported: 100rel,timer
Min-SE: 1800
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1108041665
Contact: <sip:10.31.20.14:5060>
Call-Info: <sip:10.31.20.14:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 5535 8577 IN IP4 10.31.20.14
s=SIP Call
c=IN IP4 10.31.20.14
t=0 0
m=audio 31922 RTP/AVP 0 101
c=IN IP4 10.31.20.14
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

This featurette (CSCdz58191) allows basic calls to be made between SIP and H.323 based networks, that are connected through the PGW 2200. It supports basic voice calls onlyno services from remote endpoint(s) or PGW 2200s. The following known issues exist:
Passing of DTMF does not work.
T.38 FAX does not work.
Call flows that involve receiving a SIP Re-INVITE do not trigger an H.323 ECS Invocation.
Call flows that involve receiving an H.323 ECS do not trigger the PGW to send a SIP Re-INVITE.
INAP redirection commands are not supported for SIP-H.323 or H.323-SIP calls.
http://www.cisco.com/en/US/products/sw/voicesw/ps1913/prod_release_note09186a008022f692.html#wp770756

USE OF CRM MIDDLEWARE AND XI in SRM LANDSCAPE

Hello All,
Please Can anyone tell me for communicating between SRM and different systems within a landscape,how do we recognise whether to opt for XI or CRM middleware
(Of SRM)???
Please Help.
Thanks,
Disha.

Hy Disha,
The different communication methods from SRM are:
1 CRM Middleware
This is only used for Master Data replication (init) with SAP Backends, and for Product master for init and delta uploads.
You can define your own BDocs for your needs (non easy way)
2 RFC
This is used for R/3 backend document creation (Reservation, PR and PO) in classic and extended scenarios, and for several checks in the SC creation
3 ALE/IDOCs
This is used for Good receipt and Invoice creation in R/3 in Classic and Extended scenario
4 XI
It is used for SUS integration, CCM upload and publishing scenarios, and PO send scenarios.
Regards.
Vadim

Avaya - CUCM -Meeting Place

Hi,
The current integration is between Avaya and CUCM using SIP Trunk, calling from Avaya to Cisco is working fine, however when the calls originate from the Avaya to Meeting place which uses the SIP trunk between Avaya and CUCM the call does not connect. From the logs I can see that 503 Service unavailable which normally means that the DN is not registered to CUCM.
Is there any specific configuration for integration of Avaya with Meetingplace using CUCM.
Thanks,
Vinay

First verify that the Avaya system is passing the MP number via the SIP trunk to CUCM. Then checked what number the Avaya system is passing and if that number is routed to MP. The Avaya system may be translating the MP phone number so it is not being reconized on the CUCM system.

Speed Dial and Abbreviated Dial

I have to configure 40 phones with 100 Speed dial/Abbre. numbers. Is there anyways I can configure one phone with these numbers and have this phone publish these to other 39 phones?
OR is there a easy way of doing this task?
I am using CCM 9.x and Cisco 7821 SIP phone. Any help would be greatly appreciated.

Hi
You could configure one phone, and then export the phone config (using BAT import/export functions) then edit the output file to copy the speed dials to the other phones.
On 10.5 you can associate all the phones to one user, and the User Portal allows you to link the phones together so the speed dials are synced. Just found this the other day, but it's not much use to you just now!
Aaron

Audio bridge times out

I'm trying to pull my phone conference line into a Connect webinar. Although I have set up the audio bridge with an audio profile, the Test and meeting connections both fail with a 'connection timed out' error. This often happends nearly immediately, so I don't suspect an error with the dialing or pauses. How do I troubleshoot this? Might there be an incompatibility issue with my conference line and Connect, or a limitation in my account?

thanks for the note.
No, I'm not using an integrated provider. I didn't see SIP mentioned in the setup instructions, and now I see that it is some sort of internet-enabled capacity. My conference line provider doesn't mention that in its specifications or on its help line.
So I would conclude that not just any conference provider can be plugged into the audio profile. Is that correct?

How to configure AutoAtendant on Unity Connection

Hi,
I have a Cisco CallManager 8.6 and a Cisco Unity Connection 8.6 on VMWare.I purchased the bundle (solution) Cisco BE 6000.
I did the integration of the Unity Connection with the CUCM via SIP (I also created a trunk SIP on CUCM to access CUC).
I would like to use the Unity Connection as AutoAttendant.
To do this I saw in the forum that I have to configure:
- a CTI Route Point with a DN on CallManager.
- To do a "forward all" on CTI Route Point to the VoiceMail
- On CUC to Add a new System Call Handler and to assign the same DN than the CTI Route Point
- to go to the Call Handler and to change the default message via My Personal Recording.
it would be well?
would I also have to create a end user on CUCM with Associated Device the CTI Route Point to see this DN as User with Voice Mail on CUC?
I would like that the AutoAttendat after Greeting to transfer the call to the DN which is the Pilot of Attendant Console. This action I don´t know how to do it.
I'm confused because I see on Call Handler a field called "After Greeting" with some settings (Call Action, Call Handler, Directory Handler, Conversation, User with Mailbox) but I don´t see any parameter where I can configure the DN to transfer the call.
I also the field called "Call Transfer Rules" but I am confused.
Another thing, could I to configure the CUC for the Auto Attendant works in one way some days (office hours) and out office hours and holidays works in other way?
It is possible to do on CUC to upload "Recorded Prompts" as UCCX Express to configure them as Greetings or I have to record My Personal Recordings with a Phone?
Can you help me, please?
Thank you very much

Hi,
I continued doing tests but I don´t get that the call is transfered to the Attendant Console Pilot after Greeting.
Sometimes I think that the Unity doesn´t know how to get to Atendant Console Pilot Point.
I read in the forums that I need to configure a call system transfer, by default transfer numbers that are recognized are only elements from CUC and to dial any other number which is not configured on CUC you need a call system transfer.
I also that I need a caller input or a routing rule to reach a call system transfer.
But in my case I want that after greeting the external call (DID) goes to Attendant Console Pilot without dialing any key, ie the outside caller call and he gets the greeting and after is redirected to the Attendant Console Pilot. Therefore I think I don´t need to configure any "caller input".
I did some tests with routing rule but I had not any luck. Could be due to the Unity Connection doesn´t know to get the Attendant Console Pilot.
Remember that I did the integration between CUCM and CUC via trunk SIP. How the Unity Connection knows that to get the Attendant Console Pilot it has to go to the CallManager (via SIP)?
I am confused.
Can you tell me how can I do this?
Regards

ICM with CVP, failed to find networktrunkgroup for trunkgroupid 200

Hi All,
i am trying to build ICM with CVP. i have ICM and CVP on seperate box. all pims are active everything seems to be normal at my end but for some reason call is not working.
when i dial the number on the vrupim i am getting a messge:
Trace: VRUPeripheral::RequestInstruction: Failed to find NetworkTrunkGroup for TrunkGroupID 200
and on my router i am getting the message:
**** CVP HANDOFF.TCL: AFD9EF57.89D711E0.804C80F9.53B25ADC abnormally disconnected with code 38.
please advise how to fix it.
Thanks
Munish

hi
Can you share screen shot of NTG & which version of CVP u are using . have you created Network VRU Labels.
SIP Call With IVR Service is Terminated With Reason Code: Q.850;Cause=38
Symptom:
Any failure of the bootstrap VoiceXML Server fetches to Call Server causes the SIP IVR Service
leg of call to be terminated by the recovery handoff TCL scripts on VoiceXML Server Gateway
with a Q850 code of 38.
Message:
Q.850;Cause=38
Cause:
If the Unified CVP loses network connectivity, the VoiceXML Server Gateway is not able to
get information from the IVR Service, and as a result a code 38 rejection is generated in the
Gateway logs.
remove the following lines from the network interface config in IOS:
Action:
no ip route-cache cef
no ip route-cache
no ip mroute-cache
keepalive 1800
no cdp enable
This I got it from CVP 4.x Troubleshooting guide. http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/customer_voice_portal/cvp4_0/troubleshooting/guide/cvp40tsg.pdf
and check also
http://developer.cisco.com/web/cvp/forums/-/message_boards/view_message/1726816

CVP call server service?

How call server service(ICM,SIP,IVR) are intercommunicating. i was gone through comprehensive call flow, when call hits on CVP with help of SIP service and communicate with ICM using ICM service.
HOw ICM and SIP service are communicating.

Hi Bala,
The call server uses a central messaging bus to allow each service to communicate. All the messages between these services passes via Message Bus.
Regards,
Senthil

CCM and CME integration, SIP

Hello,
This question probably was asked many times, but I'm still not able to figure out how to setup a SIP trunk between CCM (v.4.1) and CME. I read all articles about IP-to-IP SIP protocol, played around with different setting, but no luck. Can you point me to the right direction? Example configuration will be very helpful. Basically I need to make calls between 4 digits extensions between offices and to route OffNet calls.
THANK YOU!!!

Hello,
Thank you for your help. I was able to create the trunk and I'm able to make calls between CCM and CCME.
Here is my config:
SIP trunk on CCME router:
dial-peer voice 20 voip
destination-pattern [area code]80823..
session protocol sipv2
session target ipv4:10.181.0.200
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
The Trunk on CCM is simple:
- Call Classification - OnNet;
- MTP - checked;
- Destination Address - IP on CCME;
- ports - 5060, 5060, TCP, 711ulaw;
- significant digits - 4;
Thanks!

Integrating CCM 4.1(3) w/ a SIP IVR

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