Out bound using sip dialer for CVP

Hi,
I'm doing transfer to ivr treament. i done config. translation part like creating service and n/w trunk group and pheriphal no that is same in CVP -icm tab pre -routed call id: And then i added DNIS in cvp which is creted at icm transla. part.& point the DN to Voice G/w In script editor i used trnaslation route to vru.the call hitting the translation route and get released
Would able to answer need to create translation route using wizard ...Give some valuable guide to finish the lab successfulll

Hi,
can you please post a screenshot of your ICM and CVP config.
Also, does the call hit the voice gateway? What happens if you do debug ccsip mess and debug ccsip eve?
G.

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Why do we need MTP in the SIP trunk for CVP warm transfers

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Why do we need to enable MTP in SIP trunk between CUCM and CVP for CVP based trasnfers???
Thanks in advance!!
Regards,
Thammaya Gupta K.

I saw also in the CDR logs that the IP Phone media transport going to CUBE is in G711.And as well in the wireshark capture of the IP communicator that the CUCM invoke to use the g711 codec but as per ITSP logs they are now in the g729.
@ Jamie If I un-tick the MTP point required in SIP trunk will make the call leg from IP Phone to CUBE g729 (w/o hw resource), I have also tried to use g729 preferred originating codec, but still the IP Phone is using g711.
I have seen a documentation states:
" To configure G.729 codecs for use with a SIP trunk, you must use a hardware MTP or transcoder that supports the G.729 codec." - I read this on the CUCM help page under configuring SIP trunk setting.
Our ultimate goal is to use g729 without using HW MTP/ transcoder.
IP Phone ->CUCM SIP Trunk ->CUBE-> ITSP

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SIP DIALER in UCCE 9.0

Hi All,
    Can anyone explain Dial Plan configuration in SIP Dialer for Previrew and Preective.
In Predective:
       I can see SIP Dialer sends INVITE to VG, after CPA analysis Dialer sends REFER MSG to VG connect to particular EXTN and Dialer Auto answer the transferred call thriough CTI SRV.
In Preview:
     VG receives invite from CUCM and i checked in Dialer logs i couldn't see any invite, Later i blocked the translation pattern in CUCM which points towards VG and changed the agent state as Ready.
Immediatley i recieved NETWORK Error in CTI, Aftwr which i unblock the same calls dialled out sucessfully.
Please let me know for preview we need to maintain Dial-Plan in CUCM for SIP DIaler.
SIVANESAN R

Yes, You are right, But when i checked the logs for Preview VG getting Invite from CUCM,
Below is the log snippet
Predictive:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.50.43:58800;branch=z9hG4bK-d8754z-f3507a0b887a5e61-1---d8754z-;rport
Max-Forwards: 70
Require: 100rel
Contact:
To:
From: ;tag=76706749
Call-ID: d02f3c66-10694079-660e9012-32631868
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS
Content-Type: Multipart/mixed;boundary=uniqueBoundary
--More--
Supported: timer, resource-priority, replaces
User-Agent: Cisco-SIPDialer/UCCE8.0
Content-Length: 530
Remote-Party-ID: ;party=calling;screen=no;privacy=off
Preview:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.50.47:5060;branch=z9hG4bK5257890c5c
From: ;tag=714~c48d7415-a474-4367-ae32-b7af4e6f3894-29342963
To:
Date: Fri, 02 Aug 2013 21:11:04 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
--More--
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: ;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 0152463744-0000065536-0000000254-0791849152
Session-Expires: 1800
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Max-Forwards: 70
Content-Length: 0
SIVANESAN R

Max Attempts in SIP Dialer

I want to check number of Max attempt done by Dialer to connect a Phone, can yo u please guide which table exactly refer this?
w r using SIP Dialer and in configuration, we have configured 3 Max Attempts
Regards,
Hina

Hi,
there are several ways:
1. (preferred): use the Dialer_Detail view of the HDS database, looking up the number and counting the rows, e.g.
SELECT COUNT(*) AS [attempts], Phone FROM _hds.dbo.Dialer_Detail GROUP BY Phone
You may want to use a WHERE clause to filter out rows by the CampaignID and/or Date etc.
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DL__ like DL_5001_5002 where 5001 would be the Campaign ID and 5002 the Query rule ID.
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Latency in SIP Dialer

Hi All - What is the latency SIP Dialer supports (200 ms+ is this solution work if we use SIP dialer).
SIVANESAN R

Hi,
there are several ways:
1. (preferred): use the Dialer_Detail view of the HDS database, looking up the number and counting the rows, e.g.
SELECT COUNT(*) AS [attempts], Phone FROM _hds.dbo.Dialer_Detail GROUP BY Phone
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2. (usable): try to figure out the name of the dialing list table in the BA database (on the Logger A server).
DL__ like DL_5001_5002 where 5001 would be the Campaign ID and 5002 the Query rule ID.
This table actually contains the number of calls, see CallsMadeToZone1 and/or CallsMadeToZone2 - some limitations apply.
G.

How to set up SIP Proxy for comprehensive CVP and ICM mode

Hello folks,
My name is Eric and I'm facing some issues configuring a environment using the components below:
- CM 8.6
- ICM 8.5
- CVP 8.5
- SIP Proxy 8.5
- VXML GW
First of all, I always implement comprehensive mode environment at my customers (using CM, ICM, CVP, GW), but this is the first time that I'm using SIP Proxy, so the workflow change a lot. I've configure everything fine following Cisco's guides, my VRU PG's are ACTIVE, all the CVP services are UP, and the communications between the components are ok. Now I will explain below the workflow of the call:
1 - When I call to a Route Pattern 9001, for exemple, this RP use a SIP trunk that sends the call to SIP Proxy;
2 - At SIP Proxy Server I've configure a 9001 number in Route Table to send this dn to CVP Call Server;
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4 - CVP Call Server send this informations to SIP Proxy;
5 - SIP Proxy sends it to VXML Gateway and I can see this calls reaching the VXML GW;
6 - At VXML GW I've configured the dial-peer to reach my label (1234567890) and to calls the bootstrap service;
The big problem is that the calls becomes mute after being answered, I think the communication between CVP -> SIP Proxy -> VXML GW are missing some steps because if I configure my CVP to work with ICM only (excluding the SIP Proxy Server) the calls works fine following the Local Static Route and goes to available agent or queue.
If I'm using SIP Proxy Server, it's necessary to use Local Static Route or not at CVP? I think not because SIP Proxy will use the Route table to send the informations to the components. Another doubt is about ICM, for ICM is clear because the configuration is the same for use SIP Proxy or not, isn't?
Thank you very much.

Amir,
I'm not using Ingress Gateway, I'm passing from CM Route Pattern to SIP Proxy, using SIP trunk.
Saeed,
- is your media file or the agent call, which one is silent or both.
The media file is not prompted, so the calls becomes mute.
- have you configured the routes for VXML VG in route table.
Yes.
- have you configured the rotues for UCM for agent's extension in route table.
Yes
When I call, I can see it reaching the VXML gateway, but doesn't prompt the audio and neither send the call to agent. The agent becomes reserved but the call are not delivered.

Out Bound Function Module For IDoc Types CODCMT01

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Can you please tell me the Out Bound Function Module For IDoc Types CODCMT01,CODCAGN01,PRCDOC01.
Thanks..
Debi.

Hi Debi,
You can use FM MASTER_IDOC_DISTRIBUTE to distribute outbound IDoc CODCMT01, CODCAGN01 and PRCDOC01.
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HI GURU'S WISH FUNCTION MODULE USED IN IDOC'S OUT BOUND

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Hi!
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Sip dialer port export option grayed out in configuration manager ucce 10

Hi,
I am configuring the outbound option for ucce 10, when i configure the sip dialer port (dialer conifguration) in conifiguration manager, i just added 5 ports, i can see this ports showing in dialer process on the PG, but i need to export this ports to Unified communication manager, i could not able to export it because this option grayed out (i did select all, or individual selection both are having export option grayed out)
Any idea?
with Regards,
Manivannan

Exporting port configuration is only allowed for SCCP, in SIP Dialer Deployment its actually not needed and that why its grayed out.
//but i need to export this ports to Unified communication manager//
in Sip Dialer you don't need to create any Port For Dialer under CUCM. its only required in SCCP.
Sip Dialer port Directly registers with VG.
regards
Chintan
~please rate if helpful

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Thanks in Advance.
SR

Hi
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U can try to use the function module NREL_GET_NEIGHBOURHOOD
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IS_OBJECT-OBJTYPE = <business object> (I think IDOC)
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Out bound using sip dialer for CVP

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