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SIP Dialer- TRansfer to ivr

I am using Sip dialer and want to configure transfer to IVR for AMD callls. I was following dockwiki to do this as I can't see any information in anywhere else. Link I am using is:

It is asking me to create a translation route in the script and my call is failing exactly on this translation route. Any one know a better document or the steps needed to do this.

Ahmed Khalefa

Actually , Yeah .. I have this working on ICM v8.03 with CVP v8.01 ..

Just download it from here :

Hi Ahmed,

this file

is no more availble, could you please post it or share it in another way?

Thanks in advance,


Hi Pietro,

The file server is no longer available ...

I am attaching the file here .. have a good day ..

Thanks ,

Ahmed Salah

Ahmed, thanks for the document.

I must have missed it but could you point where to configure DNIS 7778889999? Is it in CVP or ICM? It doesn't say in the document.

Thanks a lot

Gergely Szabo


did you get it working?

You'll have to set up Translation routing for the Dialer as well.


Sent from Cisco Technical Support iPad App

Yeah , it is working properly ...
i did setup a Translation Routing between the Dialer to the CVP ...


I'm also facing this same issue. I have configured the Translation Route in the Script, the call are not getting dialed from the dialer to the VG. In the VG  i have configured the Dialpeer with "Rel 100" commands. Still the dial out is not happening.

Kindly provide your inputs and suggestions to reslove this issue


here is my setup:


1. Create a new Call Control Group on each IVR. Give them unique Group ID's.

2. Create a new ICM Translation-Routing application on each IVR. Create triggers.


1. Create a new Network Trunk Group. Call it NTGD . Add two Trunk Groups, one for each VRU. Add Trunks to each Trunk Group. Remember, each Trunk Group's Peripheral Number should be unique and must be equal to the Group ID set on the IVR for the Call Control Group. The Trunk Numbers are the CTI Ports from the Call Control Group created on IVR's.

2. Create two Services, one for each VRU. Add routes to each Service.

3. Using the Translation Route Wizard, create two Translation Routes, where the routing client would be the Dialer peripheral, sending calls to the NTGD (created above) Network Trunk Group (and mapped by the above Services). The labels are from the range you set for the ICM Translation-Routing application on the IVR's.

4. Set up Administrative Scripts - follow the documentation.

5. Set up a Routing Script, using the above created Translation Routes.

Gateway (only the relevant parts):


voice service voip

ip address trusted list


allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323



dial-peer voice 2001 voip

description SIP REFER -> SUB-C

destination-pattern .T

session protocol sipv2

session target ipv4:

voice-class sip rel1xx supported "100rel"

no voice-class sip reset timer expires 183

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad


dial-peer voice 2002 voip

description SIP REFER -> SUB-B

destination-pattern .T

session protocol sipv2

session target ipv4:

voice-class sip rel1xx supported "100rel"

no voice-class sip reset timer expires 183

dtmf-relay rtp-nte h245-signal h245-alphanumeric

codec g711ulaw

no vad


Notice: the IP addresses at the line starting session target are CUCM's.

Also, the version should be 15.1 (ours is (C3825-SPSERVICESK9-M), Version 15.1(3)T2)


1. Create a SIP trunk with the destination address of the VG.


Hi Gergely,

my IP-IVR campaign doesn'work properly: when the call is answered, the routing script starts but the Translation Route to VRU node fails. We have a Sip Dialer in UCCE 8.5 environment. The same Translation Route to VRU Node works ok when the script is activated by a pgUser' DN. The only change I made to my old Translation Route to VRU, was the addition of the label referred to Dialer Routing Client; this label correspond to the unique Route Point for IP IVR Translation Routing Application (71010 is tied to the IP-IVR User).

I don't understand why it doesn't work from DIaler IP-IVR Campaign DN. On the Router Event Viewer I can see the following message:

Transaltion route timeout for controller FDMPGPP_CM1 (ID 5000), route routeTransRoute (ID 5000), CRSCallID(150217,235).

If I understand it, the Translation Route is a classical one that is, is referred to the PG with the IP IVR pim: in my case I have both the pim of CUCM and IP IVR but, obviously, the pim Dialer is under another Logical Controller.

What I'm doing wrong?? Please help me... Below you can see my TR to VRU configuration.

Have a good day,



I never tried this kind of setup, although it seems logical.

Anyway, can you just try to set up a completely new translation route, with new DNIS's and Network Trunk Groups and all? Just create a regular Translation Route as if you were doing it for CUCM -> IVR translation routing, but this time it would be DIALER -> IVR. Replicate everything.

I've got two different sets of Call Control Groups on IVR (one for CUCM, one for dialer), Network Trunk Groups in ICM, Translation routes etc. Our customer wanted to ensure there would be a dedicated set of IVR ports for Dialer, so I did not even think of having one NTG for both CUCM and Dialer.


Thanks a lot Gerely... you are very fast ;-)

I'm thinking that perhaps could be a Gateway/CUSP configuration: when TR to VRU node send the label 71010 to the CUCM and after to the IP-IVR, it passes through the sip trunk and then some Dial peer has to match for the 71010 dn too.

I believe that the problem is there, on the gateway or in the CUSP.

Now, my telephonic collegues are gone away... on moday I'll try to modify the dial peers.

Thanks a lot again Gergely ...

Have a nice week end,


Hi gerely,

Sorry for my  delayed response. Your setup was very helpful and calls started dialing out but were getting dropped before answering and to fix the issue, was suggested to added the command   " progress_ind callproc enable 8" on the dial-peer on the gateway to allow successful ringback passing from PSTN to SIP.  After this calls started to reach destination phones successfully but were disconnected after going off hook. Suspecting the issue on the Codec but got to confirm the behavior.

Thanks again!!!