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

Similar Messages

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

UCCE SIP Dialer not active and PIM neither

Dear Networkers,
We have UCCE 8.5(4) installed.
We have configured and installed aSIP dialer. The issue is that the SIP dialer is not working and the PIM is not active.
From the Dialer process output I can see the following after enabeling the EMSDisplayToScreen :
18:02:48 Trace: Trying to connect to server: 17.20.10.80 on port: 44027
18:02:48 Trace: socket open on Server=17.20.10.80 port=44027
18:03:09 Trace: Error calling Connect(): 10060
18:03:09 Trace: Could not connect to either CTI Server A or CTI Server B
18:03:09 Trace: CTI Server configuration:
18:03:09 Trace:         CTI Server A: 17.20.10.80, Port: 44027
18:03:09 Trace:         CTI Server B: 17.20.10.80, port: 44027
From the MR PIM process output I can see the following after enabeling the EMSDisplayToScreen :
18:03:40 ADDED\UPDATED 0 PeripheralTargets on Peripheral 5008.
18:03:40 ADDED\UPDATED 0 PeripheralMonitors on Peripheral 5008.
18:03:40 ADDED\UPDATED 2 MediaRoutingDomain on Peripheral 5008.
18:03:40 Trace: Transport: appServerHostName1 = kbprcrmipc01a01
18:03:40 Trace: Transport: appTcpServiceName1 = 38003
18:03:40 Trace: Transport: appServerHostName2 =
18:03:40 Trace: Transport: appTcpServiceName2 =
18:03:40 Peripheral 5008 sending OPC PIM_OK_ACK acknowledgment for command PIM_CONFIGURE_REQ (TransID=0).
18:03:40 Attempting to connect to MR application at IP address 172.20.10.80 port 38003.
18:03:50 Attempting to connect to MR application at IP address 172.20.10.80 port 38003.
18:04:02 Attempting to connect to MR application at IP address 172.20.10.80 port 38003.
18:04:11 ProcessPIMSetIdleReq: Peripheral 5008 going idle.
18:04:14 Peripheral 5008 sending OPC PIM_OK_ACK acknowledgment for command PIM_SET_IDLE_REQ (TransID=33).
18:05:11 ProcessPIMSetIdleReq: Peripheral 5008 going idle.
18:05:11 Peripheral 5008 sending OPC PIM_OK_ACK acknowledgment for command PIM_SET_IDLE_REQ (TransID=34).
For the Dialer process, I can see that the dialer cannot connect the CTI Server on the port 44027, but the CTISVR is active and working fine (CTIOS also).
For the PIM process, I can see that it couldn't connect to the dialer on the port 38003, but the configuration is correct.
Any help please ?
Thanks in advance.

Hi haouaswajih,
After referring to the below link,
http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/outbound_option/outboundoption9_0/installation/guide/UCCE_BK_O4A87BBC_00_outbound-option-guide-for-cisco_chapter_0100.html
Step 10
On the last Outbound Option Dialer Properties dialog box , specify the following        information:
Outbound Option server: The host name or IP address of the Outbound           Option server. This server is typically the same machine where the Outbound Option           Campaign Manager is located.
CTI server A: The host name or IP Address of the machine that has           side A of CTI server installed.
CTI server port A: The port number the dialer uses to interface with           CTI server side A. The default is 42027.
CTI server B: For duplexed installations, the host name or IP Address           of the machine that has side A of CTI server installed.
CTI server port B: For duplexed installations, the port number the           dialer uses to interface with CTI server side B. The default is 43027.
Heart beat: How often the dialer checks its connection to the CTI           server, in milliseconds. The default value of 500 is acceptable.
Media routing port: The port number the dialer uses to interface with           the Media Routing PIM on the Media Routing PG. The default is 3800 to 3801.
Call Manager TFTP server: The host name or IP address of the           CallManager TFTP server. This server is the same machine used for the CallManager           publisher.
Could you please crosscheck the CTI ports being used\pointed here in your setup.
Hope this helps,
Anand
Please rate helpful posts !!

UCCE 10.5: MR PG can not connect to SIP Dialer

Hello,
I have a strange problem with a SIP Dialer installation.
The MR PIM can not connect to SIP Dialer with the following error:
14:32:34:621 pg2a-pim1 Attempting to connect to MR application at IP address: 172.19.10.31, port: 38001.
14:32:34:621 pg2a-pim1 Trace: Transport: Making connection attempt to host1: 172.19.10.31 port: 38001.
14:32:35:713 pg2a-pim1 Trace: Transport: Connection to host1 attempt failed; error 10061..
14:32:35:713 pg2a-pim1 Trace: Transport: The attempt to connect was rejected..
14:32:35:713 pg2a-pim1 Trace: Transport: Make sure that the MR-PIM setup has the correct MR application hostname and port number..
14:32:35:713 pg2a-pim1 Trace: Transport: Make sure that the application is listening for connection at the correct port number..
14:32:35:713 pg2a-pim1 Trace: Transport: Connection closed..
I configured two duplexed SIP Dialers without Port Map. In baDialer logs I can see it can connect to Campaign Manager, CTI Servers, but it received "Port Map With ZERO Ports". Judging by what is written in SRND, this behavior is normal.
14:38:51:052 dialer-baDialer Trace: DialerMetrics::reset(), Dialer Metric subsystem was reset..
14:38:51:350 dialer-baDialer Initializing Event Management System (EMS) library.
14:38:51:350 dialer-baDialer EMS library initialized with write thread.
14:38:51:350 dialer-baDialer Trace: EMS Server pipe cce\Dialer\baDialerEMSPipe enabled for cce\Dialer\baDialer.
14:38:51:350 dialer-baDialer Trace: Release 10.5.1.0 , Build 2572.
14:38:51:350 dialer-baDialer Trace: Received dialer startup request..
14:38:51:350 dialer-baDialer Trace: Monitor Server pipe cce\Dialer\baDialerCmdPipe enabled for cce\Dialer\baDialer.
14:38:51:351 dialer-baDialer Trace: Dialer starting...ICM\cce\Dialer.
14:38:51:351 dialer-baDialer Trace: Using registry key: Software\Cisco Systems, Inc.\ICM\cce\Dialer.
14:38:51:351 dialer-baDialer Trace: Using registry key: Software\Cisco Systems, Inc.\ICM\cce\CurrentVersion.
14:38:51:354 dialer-baDialer Trace: EMT I/O completion ports: max threads=4, concurent threads=0.
14:38:51:355 dialer-baDialer Trace: Current Directory (C:\icm\cce\dialer).
14:38:51:355 dialer-baDialer Trace: Size of call result mapping table: 46.
14:38:51:355 dialer-baDialer Trace: DialerMetrics::Instance, Dialer Manager Creating New instance..
14:38:51:355 dialer-baDialer Trace: The performance monitor object instance (cce) was created successfully..
14:38:51:359 dialer-baDialer Trace: Attempting EMT connection to computer [172.19.10.30/172.19.10.30], port [40032].
14:38:51:611 dialer-baDialer Trace:    EMT connection established.
14:38:52:368 dialer-baDialer Trace: Registering with Campaign Manager.
14:38:52:370 dialer-baDialer Trace: Config received: [1] dialer PortThrottle:15.000000.
14:38:52:370 dialer-baDialer Trace: Trying to connect to server: 172.19.10.31 on port: 42027.
14:38:52:370 dialer-baDialer Trace: socket open on Server=172.19.10.31 port=42027.
14:38:52:370 dialer-baDialer Trace: (CTIPROXY)   Sending CTI OpenRequest for this Dialer.
14:38:52:372 dialer-baDialer Trace: CallListener creating mutex - Global\CCallListener8536.
14:38:52:372 dialer-baDialer Trace: (PM) Update Port Throttle. lPortThrottle=15.000000, dwPortThrottleCount=30, dwPortThrottleTime=2.
14:38:52:372 dialer-baDialer Trace: Received Port Map With ZERO Ports.
14:38:52:372 dialer-baDialer Trace: Received Port Map With ZERO Ports.
14:38:52:372 dialer-baDialer Trace: Received Port Map With ZERO Ports.
14:38:52:372 dialer-baDialer Trace: Received Port Map With ZERO Ports.
14:38:52:372 dialer-baDialer Trace: (CLMGR)   CTI Client Event Report Request, invoke: 1, clients: [1].
14:38:53:382 dialer-baDialer Trace: Dailer Status Change,Old Status:cce-Dialer BADialer_SIP -X [CM-X] [CTI-U] [Ports-X] [MR-X] [SIP-U] 0x0.
14:38:53:382 dialer-baDialer Trace: Dailer Status Change,New Status:cce-Dialer BADialer_SIP -X [CM-A] [CTI-A] [Ports C:0,R:0,B:0] [MR-X] [SIP-U] 0x3.
14:38:53:382 dialer-baDialer Trace: (DD)   **** Media Routing PIM is not Connected ****.
14:38:53:382 dialer-baDialer Trace: (DD)   **** All SoftPhone Channels not Initialized ****.
14:38:53:382 dialer-baDialer Trace: (DD)   **** Configured SoftPhone Channels: [0], Initialized: [0] ****.
14:38:54:396 dialer-baDialer Trace: (DD)   **** Media Routing PIM is not Connected ****.
14:38:54:396 dialer-baDialer Trace: (DD)   **** All SoftPhone Channels not Initialized ****.
14:38:54:396 dialer-baDialer Trace: (DD)   **** Configured SoftPhone Channels: [0], Initialized: [0] ****.
14:38:55:410 dialer-baDialer Trace: (DD)   **** Media Routing PIM is not Connected ****.
14:38:55:410 dialer-baDialer Trace: (DD)   **** All SoftPhone Channels not Initialized ****.
I double checked the IP Addresses, MR port(38001), network interfaces order in Windows and everything looks ok.
A possible problem could be that I see no application listening on port 38001 with netstat -na.
Does anyone have any idea what else should I check?
Many thanks in advance,
Cristian

Also, here is an output of the Diag Portico when your MR is up and ports are registered and everything is healthy

SIP Dialer Gateway

Hello,
I am building a design for one of our customers who is intersted to have the SIP dialer.
I read about the SIP dialer and it looks like that the rel1xx on the SIP config should be enabled but for the CVP it needs to be disabled.
Is this mean that i have to had a sperate GW for the SIP dialer or there is some workaround for this.
Thanks
Amer

Amer,
Everyone is right... You need one Dial-peer for the SIP Dialer w/ REL1xx enabled, however what was not mentioned is you cannot hand the call off to CVP with out a translational routing. Your CVP Dial-Peer will actually need to be configured to match the did range you defined for tranlation routes, in my environment it 12[1-2].. for my first CVP Call Server and follow suit for the remaining 3. When the Dialer hands the call off to CVP it will use the trans route which will need to match a a Dialpeer for the CVP Call Server as the Voice Gateway will be the one performing the actual transfer.
Also see my post regarding Dissabling Ring Back for SIP Dialer Calls.
https://supportforums.cisco.com/docs/DOC-21998
Make sure you adhere to ALL the IOS revision requirements defined in the UCCE SRND and CVP SRND ** A2Q(Cisco BU) will will not certify the design otherwise and if they overlook the IOS versions then TAC will definatly tell you the first time you open a TAC case that you have to modify the IOS version to match the SRND.
My CC
CVP 8.5
ICM 8.5.2
CUCM 8.5.1
CUSP 8.5
IOS 15.1.3T2

Sip dialer is not active and mr pim

i installed call server--> router,ctoserver,ctios,pg,in pg1 agent pim, vru pim, in pg2 mr pim and dialer
data server -----. logger, aw
this is pcce setup details
agent pim,vru pim, cti server all servers are active
issue is mr pim is not active, dialer is allso not active,
mr pim logs
09:20:06:251 PG2A-pim1 Peripheral 5010 sending OPC PIM_OK_ACK acknowledgment for command PIM_CONFIGURE_REQ (TransID=0).
09:20:06:253 PG2A-pim1 Attempting to connect to MR application at IP address 192.168.11.10 port 38001.
09:20:17:272 PG2A-pim1 Attempting to connect to MR application at IP address 192.168.11.10 port 38001.
09:20:28:296 PG2A-pim1 Attempting to connect to MR application at IP address 192.168.11.10 port 38001.
09:20:37:257 PG2A-pim1 ProcessPIMSetIdleReq: Peripheral 5010 going idle.
09:20:40:263 PG2A-pim1 Peripheral 5010 sending OPC PIM_OK_ACK acknowledgment for command PIM_SET_IDLE_REQ (TransID=2).
09:20:40:267 PG2A-pim1 Peripheral 5010 sending OPC PIM_OK_ACK acknowledgment for command PIM_CONFIGURE_REQ (TransID=0).
09:20:40:269 PG2A-pim1 Attempting to connect to MR application at IP address 192.168.11.10 port 38001.
09:20:51:294 PG2A-pim1 Attempting to connect to MR application at IP address 192.168.11.10 port 38001.
09:21:02:316 PG2A-pim1 Attempting to connect to MR application at IP address 192.168.11.10 port 38001.
09:21:10:724 PG2A-pim1 ProcessPIMSetIdleReq: Peripheral 5010 going idle.
Dialer:
09:11:20:531 dialer-baDialer Trace: Attempting EMT connection to computer [192.168.11.11/192.168.11.11], port [40072]
09:11:21:531 dialer-baDialer Trace: Unable to establish an EMT connection:
09:11:21:531 dialer-baDialer Trace: error value [10060], text: A connection attempt failed because the connected party did not properly respond after a period of time, or established connection failed because connected host has failed to respond

MR PIM id 5010, ip address 192.168.11.10, port number 38001, private and visible router interfaces is 192.168.11.10(call server)
in dialer ip 192.168.10.240, sip service port 5060, compain manager 192.168.11.11, cti server is 192.168.11.10, cti server port:42027, midia routing port:38001,
if i install dialer in data server dialer is active but mr in is idle, but in document sepcification should be install call server only, so i removed from data server again i installed in data server,
so i am facing sip dialer and mr pim not in active
i see pgagent is active, but pim is not in active

SIP Dialer

•1.       Can we integrate SIP Dialer with Microsoft Dynamics CRM 2011 (Online) or not.
•2.       If Yes then is there any API available for SIP Dialer the integration with MS Dynamics CRM 2011 (Online)
SIVANESAN R

This has nothing to do with the dialer, but with your desktop. Yes, you should be able to pop dynamics, depends on the type of desktop you're using CAD and Finesse would probably the be easiest, if you have CTIOS you're going to need to custmize it.
david

SIP Dial Rule ATA187

Hi Guys
I have a problem. I´m trying to configure a SIP Dial Rule in the CUCM 8.6 , using an ATA187, but no works.
Here, with and without PLAR.
Any idea?
TIA
Cristian

Hi Robert
I performed the following steps:
Step 1 Create a partition, for example, P1, and a calling search space, for example CSS1, so CSS1 contains P1. (In Cisco Unified Communications Manager Administration, choose Call Routing > Class of Control > Partition or Calling Search Space.)
Step 2 Create a null (blank) translation pattern, for example, TP1, which contains calling search space CSS1 and partition P1. In this null (blank) pattern, make sure that you enter the directory number for the B1 PLAR destination in the Called Party Transformation Mask field. (In Cisco Unified Communications Manager Administration, choose Call Routing > Translation Pattern.)
Step 3 Assign the calling search space, CS1, to either A or A'. (In Cisco Unified Communications Manager Administration, choose Device > Phone.)
Step 4 Assign the P1 partition to the directory number for B1, which is the PLAR destination. (In Cisco Unified Communications Manager Administration, choose Call Routing > Directory Number.)
Step 5 For phones that are running SIP, create a SIP dial rule. (In Cisco Unified Communications Manager Administration, choose Call Routing > Dial Rules > SIP Dial Rules. Choose 7940_7960_OTHER. Enter a name for the pattern; for example, PLAR1. Click Save; then, click Add Plar. Click Save.)
Step 6 For phones that are running SIP, assign the SIP dial rule configuration that you created for PLAR to the phones, which, in this example, are A and A'. ((In Cisco Unified Communications Manager Administration, choose Device > Phone. Choose the SIP dial rule configuration from the SIP Dial Rules drop-down list box.)
But not works
TIA
Cristian

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.
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.

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
You may want to use a WHERE clause to filter out rows by the CampaignID and/or Date etc.
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.

SIP Dialer codec

Hello,
There is one question to community:
Where we can setup preferable codec in Cisco SIP dialer?
It is needed to change g711ulaw which is default SIP Dialer codec to g711alaw.
Thanks in advance,
Regards,
Alex

Chintan,
I see that INVITE from dialer comes with one codec g711ulaw. Please see below:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.13.81.3:58800;branch=z9hG4bK-d8754z-57145a04a377e42c-1---d8754z-;rport
Max-Forwards: 70
Require: 100rel
Contact: <sip:[email protected]:58800>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=23395838
Call-ID: eb13700c-00095a52-50032441-92184f3b
CSeq: 1 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, PRACK, REFER, NOTIFY, OPTIONS
--More--
Content-Type: Multipart/mixed;boundary=uniqueBoundary
Supported: timer, resource-priority, replaces
User-Agent: Cisco-SIPDialer/UCCE8.0
Content-Length: 608
Remote-Party-ID: <sip:@10.13.81.27>;party=calling;screen=no;privacy=off
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 2884 2524 IN IP4 172.19.155.41
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 19994 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--uniqueBoundary
--More--
Content-Type: application/x-cisco-cpa
Content-Disposition: signal;handling=optional
Events=FT,Asm,AsmT,Sit,Piano
CPAMinSilencePeriod=608
CPAAnalysisPeriod=2500
CPAMaxTimeAnalysis=3000
CPAMaxTermToneAnalysis=30000
CPAMinValidSpeechTime=112
Regards,
Alex

SIP REFER with UCCE and CUPS SIP Proxy

I am running UCCE 8.0.1 with CVP and CUPS as the SIP proxy. I am looking to transfer calls to PSTN and release from CVP to free CVP ports. I am using the rfxxxxxxx method in the ICM script, which seems to work fine from CVP. However the SIP proxy send an Invite to our SBC instead of a REFER. Is there a way to configure SIP Proxy to send the REFER instead of the invite? I would like to release the call from our SBC as well.
The other idea was to insert a custom header in CVP I could then pull out at the SBC and replace with a REFER. Does anyone have any links to documentation on this?
Thanks
TC

how about check "enable send calls to originator" for the refer label routing in CVP? this would bypass proxy.

PRI Utilization in SIP Dialer

Hi All - Is there any way to achieve Utilization of Multiple PRI in Single Gateway with CUSP Deployment.
Let's say i have 5 PRI'S in same VG, Suppose if one of the PRI goes down is there any option to utilize remaining 4 PRI'S in the same VG.
I checked SRND, where i can see IF VG is overloaded and loses it's WAN connectivity with PSTN it will return error code as 503
In CUSP we can configure Failure response code as 503 so that CUSP will take alrenate element and route it accordingy.
If that is the case, We are not utilizing remaining 4 PRI'S in the first VG.
Is there any way to achieve the remaining 4 PRI'S without jumping to alternate element in CUSP.
SIVANESAN R

Hello Sivanesan,
If you have 5 PRI's. Do you have 5 different dial-peers in the Voice Gateway for the same destination pattern ? Do you have Preference command given in the dial-peer ? Basically If a Call matches more than one outbound dial peer, the router itseld hunt them in order as per your configuation. If the Call setup failed for some reason, the next dial-peer will be attempted. Its based on the hunt order you specify.
In the voicegateway global config mode use the below command, you should see 7 options, based on your configuration the dial-peer will be hunted. This happens within the Gateway/Router before sending the response to CUSP
VG3845(config)#dial-peer hunt ?
<0-7> Dial-peer hunting choices, listed in hunting order within each choice:
0 - Longest match in phone number, explicit preference, random selection.
1 - Longest match in phone number, explicit preference, least recent use.
2 - Explicit preference, longest match in phone number, random selection.
3 - Explicit preference, longest match in phone number, least recent use.
4 - Least recent use, longest match in phone number, explicit preference.
5 - Least recent use, explicit preference, longest match in phone number.
6 - Random selection.
7 - Least recent use.
Regards,
Senthil

Patient reminder outbound sip dialer script

Hi Experts,
I have setup some UCCX 8.5 system ACD scripts with agents on CUCM 8.6 for a healthcare facility.
The only item remaining is an outbound dialer option, which can automatically call out
patient phone #s and play a recording with Patient name, date and time of appointment, etc..
On the back-end, these appointments are stored in an Oracle based Medical application.
I am trying to figure out what format I need to export the data from Oracle, so it can be imported
into the UCCX ?
here are the pieces of the puzzle:-
1> Oracle DB will export a daily/weekly appointment list, which I want to import into UCCX.
2> the hospital receptionist will record the patient names as wav files (they dont have any Nuance integration)
3> my Outbound dialer script will either import the static file (from the DB), or do a Database DIP
     to query this list of appointment reminders, then create the reminder prompt, and
     complete the call/play prompt after the call is answered....
Questions:
> does anyone have any sample script/pointer to this kind of an application integrated with UCCX ?
> what format do I need to export data from Database in ? (assuming the DB does not support a direct
   connection through UCCX)
> any other pointers/suggestions are welcome.
I searched through the support forums, but didnt find this answered before..
Thanks
JK

A CCX script can do a database dip. I am not sure if Oracle is a supported DB for CCX.
The outbound dialer is more about dialing and send calls to Agents.
What CCX is missing is a way of triggering a script at regular time intervals. However there are two ways to trigger a script in CCX. The first is by someone dialing in using a route point. The second is via a http access.
So what you could do is to schedule a job on a Windows server that every minute would make a web access to the CCX server. This would trigger your CCX script which could then do a database dip looking for a user to call. If it finds work it would dial the number, play the messages etc and possibly write the results back to the database. This is nothing to do with the outbound dialer.
Nuance is a TTS/ASR engine and is pretty simple to integrate, well TTS is pretty simple to setup and use. ASR is a bit more challenging.
If I was doing this job I would write a Windows service that ran a regular intervals. It would look in the Oracle DB for work. When it found a person to call it would make a web access to CCX to trigger a script and it would pass the information across as parameters. This would be the number to dial, date and time of the appointment, name of the wav file with the persons name and what ever information you need to pass to the script. Depending on how your script works you could have CCX return a result to the Windows service, things like no answer, wrong person, person wants to change appointment. I would do the smart things in the Windows service and keep the CCX script simple.
Graham

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.

SIP DIALER in UCCE 9.0

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