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If you just need a simple password control for long distance or international call and don't want to use RADIUS, you can set a parameter in service then in script ask caller to enter password if it matched the one set in IOS configuration then make the call out otherwise disconnect call. A sample script is posted in Documentation section.

 

Hi Cisco Developers,

MSE VOIP Sample Application along with User Guide can be found under the below location.
http://developer.cisco.com/web/vgapi/docs

The goal of this sample application is to demonstrate the Integration of IVR(Interactive Voice response) and MSE(Mobility service Engine).

MSE is a Context-Aware Mobility solution allows organizations to capture and integrate business processes with detailed contextual information about items such as location, temperature, availability of a person or whether an asset is in motion. This solution leverages the Unified Wireless Network to collect real-time contextual information about mobile assets and users. Once all the contextual information has been captured, calculated and stored by the Context-Aware software module, it can be made available to any business application that needs it through the Cisco Context-Aware Mobility API, which is based on Simple Object Access Protocol (SOAP) and XML protocol.

For more information about MSE you can refer http://developer.cisco.com/web/contextaware/home

This Sample Application provides the single/multiple mobile client location details in a wireless Network through IVR(Interactive voice Response).

just download this web application & deploy in any web server like tomcat have fun to add IVR(Interactive voice Response) functionality to your Mobility service Engine.

Regards,
Raghavendra

 

In IOS 15.2(2)T a new option is introduced for infotag get leg_dnis and infotag get leg_ani

"callnumber" is keyword

infotag get leg_dnis
infotag get leg_ani

example:

infotag get leg_dnis callnumber leg_incoming


This option services the following purposes:

1. Return number portion only for DNIS or ANI in SIP trunk
Without this option we need to use regular expression to extract the number portion after we use ¿infotag get¿ to get the ANI or DNIS
For example:
set dnis [infotag get leg_dnis]
regexp {[0-9]+} $dnis dnis
With using this option we don¿t need to use regular expression anymore. The command will return number only.

2. In certain configuration if we use ¿voip-incoming translation-profile¿ then use after translation number to trigger Tcl script, the ¿infotag get leg_dnis¿ command will return the number before translation.
For example:
The incoming number is 7771234 and we need have it translated to 8881234 via ¿voip-incoming translation-profile¿ command. We will expect 8881234 to hit Tcl application in dial-peer for incoming 8881234. Script will be triggered as expected but ¿infotag get leg_dnis¿ will return 7771234 but not 8881234. Use new option can fix this issue.

 

The fetchaudio attribute specifies the URI of the audio to be played while the document is being fetched. It can be used with any one of the following VXML Elements <data>; <goto>; <link>; <subdialog>; <submit>

Example:
<submit next="/VXML/Server" method="post" namelist="VGAPI_vxmlLog" fetchaudio="/VXML/audio/Hi.wav" fetchaudiodelay="1s" fetchtimeout="100s" />

  1. All prompts queued prior to the fetchaudio will be played prior to the fetchaudio.
  2. All prompts played before the fetchaudio are non-bargeinable.
  3. Audio associated with fetchaudio is also non-bargeinable.
  4. All prompts before the fetchaudio are guaranteed to be played to completion, regardless of when the fetch finishes. See Known issues for important information regarding this point.
  5. Audio associated with fetchaudio will be interrupted as soon as the interpreter is ready to flush the next set of prompts from the queue.
  6. When a sequence of elements specifies fetchaudio, the interpreter interrupts the previous fetchaudio when it begins to process the next element that has a fetchaudio attribute.
  7. The interpreter does not loop the audio associated with fetchaudio. If the interpreter is still processing the associated request when the fetchaudio playback has completed, dead air will result. To avoid this
  8. It is recommended that your fetchaudio be much longer than your fetch. Consider a five minute audio clip.

 

Cisco Technology Developer Partners,

Cisco is proud to announce the availability of our Cisco Unified Communications System Release 8.6 Not-For-Resale software bundle on Cisco Marketplace (Partner Login Required).

To purchase the latest Unified Communications NFR Software bundle
- Navigate to Cisco Marketplace
- Login using your Cisco.com UserID
- Click on Unified Communications NFR Software (display box)
- Select Unified Communications Not-For-Resale Products link (small link at top of page)
- Select Cisco Unified Communications System Release NFR Program (drop down menu on right)
- Select Technology Developer Partner
- Click Add to cart

PRE-ORDER: Cisco Unified Communications System Release 8.6 - Partner Bundle Offering (PBO).
Part#: UC8.6-K9-PBO Rev. 0 - Release date : Aug 19 2011
Note: Cisco Unified Contact Center Enterprise is not currently available in UC8.6 Partner Bundle. Orders will begin to ship early September.

**Please allow 1 week for Partner Verification**This Partner Bundle is not for resale
**Can only be shipped to Advanced UC Partner, Technology Development Partner and Learning Solutions Partner**

Below are the Cisco Collaboration software applications included in the offering:

Cisco Unified Communications Manager 8.6(1)
Cisco Unified Communications Manager 8.6(1) Session Management Edition
Cisco Emergency Responder 8.6(1)
Cisco Unity Connection 8.6(1)
Cisco MeetingPlace 8.5
Cisco Unified Presence 8.x
Cisco Unified Personal Communicator 8.6
Cisco Jabber for MAC 8.6(1)
Cisco Unified Integration for Microsoft Lync (CUCI-Lync) 8.6(1)
Cisco Unified Integration for Tencent Real-Time eXchange 8.6 (CUCI-RTX)
Cisco Unified Contact Center Express 8.5(1)
Cisco Voice Portal 8.5
Cisco MediaSense 8.5
Cisco Unified Communications Management Suite (CUCMS) 8.6(1)

There is no hardware included in this offering. The hardware used must match the hardware requirements of the individual components.

The contents of this kit are:
Cisco Unified Communications Manager 8.6(1): Includes software media kit for Unified Communications Manager, Unified Intercompany Media Engine
3 Node license (Standard hardware or VMware platforms), 150 Device License Units (DLU), SW Feature (supports Major version upgrades), Training Kit
License authorization key/file can be obtained using the Product Authorization Key (PAK); PAK allows upgrade from previous CUCM 8.5 NFR software
Cisco Unified Communications Manager 8.6(1) Session Management: Includes software media kit for Unified Communications Manager Session Manager Edition and Unified Communication Manager Business Edition; No license keys required.
Cisco Emergency Responder 8.6(1): Includes software media and Product Authorization Key (PAK) for Cisco Emergency Responder; License file can be obtained using the PAK for 100 user licenses and one server
Cisco Unity Connection 8.6(1): Includes software media and default licenses for 10 mailboxes, 2 ports, 30 seconds messages maximum
Cisco MeetingPlace 8.5
Includes software media for Cisco MeetingPlace
Includes server software license and 10 concurrent user licenses for audio and video conferencing
Cisco Unified Presence 8.x: Includes software media and license for single Cisco Unified Presence server
Cisco Unified Contact Center Express 8.5:
Includes software media kit and Product Authorization Key (PAK) for Cisco Unified Contact Center Express;
License file can be obtained using the PAK for High Availability with 6 Premium seat licenses
Cisco Unified Contact Center Enterprise 8.5: Includes software media kit for Cisco Unified Contact Center Enterprise, Cisco SocialMiner, and Cisco Finesse. Cisco Unified Customer Voice Portal 8.5: License PAK included. PAK is used to acquire a license file that enables up to 30 VXML sessions.
Cisco Voice Portal 8.5(1)
Includes software media kit and Product Authorization Key (PAK) for Cisco Voice Portal and VXML Server
License file can be obtained using the PAK that enables up to 30 VXML sessions
Cisco MediaSense 8.5(2): Includes server software license and 100 ports licenses for audio and video recording, all on a single machine.
Cisco Unified Communications Management Suite 8.6:
- Unified Provisioning Manager 8.6(1),
- Unified Operation Manager 8.6(1),
- Unified Service Monitor 8.6(1),
- Unified Service Statistics Manager 8.6(1)

Downloads

Cisco Unified Personal Communicator 8.5(2) Download software from http://www.cisco.com/cisco/software/navigator.html under Voice and Unified Communications
Cisco Jabber for Mac Download software from http://www.cisco.com/cisco/software/navigator.html under Voice and Unified Communications
Cisco Unified IP Communicator 8.6(1) Download software from http://www.cisco.com/cisco/software/navigator.html under Voice and Unified Communications
Cisco UC IntegrationTM Integration for Microsoft Lync 8.6(1) Download software from http://www.cisco.com/cisco/software/navigator.html under Voice and Unified Communications
Cisco UC Tencent RTX 8.6(1) Download software from http://www.cisco.com/cisco/software/navigator.html under Voice and Unified Communications
Cisco Unified Mobile Communicator Clients Download software from http://www.cisco.com/cisco/software/navigator.html under Voice and Unified Communications or from the Applications store

 

New Video Tutorial section is added for this Tcl IVR API
Video Tutorial

 

param CLI can be mapped to IVR either in the global application mode, or as group params or under the specific dial-peer.

config t
application
service aa flash: app-b-acd-aa.tcl
param aa-pilot 7001

Having the param under the specific dial-peer provides the flexibility by having same application to work with various dial-peers in various ways.
For Example the same AA script can be used with the different pilot numbers for different incoming called numbers.

config t
application
service aa flash: app-b-acd-aa.tcl
!
dial-peer voice 1 voip
service aa
paramspace aa aa-pilot 7001
incoming called-number 7001
!
dial-peer voice 2 voip
service aa
paramspace aa aa-pilot 7002
incoming called-number 7002
!
dial-peer voice 3 voip
service aa
paramspace aa aa-pilot 7003
incoming called-number 7003

Hope this helps someone.

 



Cisco Developer Network will be presenting a CDN Developer Track at Cisco Live! London the week of January 31, 2011.

We are presenting technical sessions which highlight Application Programming interfaces (APIs) and Software Developer Kits (SDKs) for Cisco technologies such as Unified Communications, IOS, and Access Routing Technologies ¿ including the new Cisco Cius Application Development SDK ! You¿ll get technical insight into the available interfaces, hear from subject matter experts and learn about the benefits of becoming a Cisco developer.

Learn more about the new Cisco Live track for the Developer Network Program!

http://www.ciscolive.com/europe/attendees/education/dnp

 

There are a few more tones can be played but are not listed in the programming guide

tn_ringback -- Plays a ring back tone
tn_congestion -- Play a congestion tone
tn_stuttdial -- plays a stutter dial tone
tn_intercept --plays a intercept tone, like busy tone

For example I have to use "leg setup $dest callinfo" for a reason
I don't provide incoming leg, so when the ev_alert or ev_progress coming in script won't be able to ring calling party.
At this moment I can use "playtone leg_incoming tn_ringback" to provide local ring back"

Hope this can help someone when they need it.

 

Ultimate AA script


Suffering the AA menu prompt tree changing that you need to modify the script frequently? Or trying to find an AA script can fit all menu prompt tree?
You really can create an ultimate script that can fit all prompt structure
Here is the solution:
Think about what is changing? It¿s the menu prompt tree. So instead of modifying script, we create a flat file that can hold all prompt tree information. If we need to change it we change that data file. So the script¿s responsibility is only to have the intelligence to read and interpret the data file and we don¿t need to modify script when menu tree changed.

Example:

#Beginning of prompt tree data file
#Should have a main menu prompt(welcome prompt) to lead called to
#the following menu tree
#first level down (Sub menu level 1)

1 1001
2 _Tech_Support.au (1 for data, 2 for voice, 3 for video)
3 3001
#second level down (sub menu level 2)
21 2001
22 2002
23 2003
# end of data file

We need a main menu prompt as usual Caller will hear this prompt when calling pilot number This is the same as normal AA welcome prompt, it will lead the caller to the menu above
We should have a configurable parameter in script for data file name so we can configure it from IOS . When script start reading the data file it should interpret the script like this:
When first character is not a number ignore that line.
When there is a number follow by the first then it¿s a destination number, do a call setup to that destination number. If it¿s an audio prompt name, play that prompt then continue. Script should keep reading the file until reach a destination number.
Theoretically it can go to any depth of prompt tree.

Use the data above as an example:


The welcome prompt will guide users to press 1 for ¿, press 2 for ¿ and press 3 for ¿.
Notice that options 1 & 3 are associated with numbers that correspond to Tcl application where you know the destination number you want to reach. Therefore, it will connect that destination number.
Option 2 is associated with another prompt, at which further information will be given to the caller until a final destination number is obtained. At the 2nd level, the first digit is ¿2¿ which signifies that it is from option 2 in the previous level
The process continues to the last level or until a final destination number is reached


Take another example

1 1001
2 _Sales.au (1 for new, 2 for used)
3 3001
# second level down (sub menu level 2)
21 _new.au (1 for res, 2 for comm.)
22 _used.au (1 for res, 2 for comm)
# third level down (sub menu level 3)
211 2001
212 2002
221 2003
222 2004

If caller select 2-1 then will hear the selection 1 for residential 2 for commercial
if caller select 2-1-1 will be connected to 2001 which is sales ¿ new -residential
If caller select 2-2-2 will be connected 2004 which is sales ¿used ¿ commercial

This is the basic concept, hope it gives you an idea to create your own ultimate AA script.

 

I would like to shed some light on IPIPGW concept and its features.
IPIPGW Over TCL-VXML scripts:
We can deploy TCL and VXML scripts on IPIPGW,where DSP¿s are not requried to use this feature.You have to record prompt in the same codec type as you use for a call.This feature will be useful when we deploy prepaid calling cards.
IPIPGW :
A GW connects 2 IP - IP call legs in called IPIPGW.Normally a TDM GW connects a telephony call leg and a voip call leg,but in IPIPGW the 2 call legs are VOIP. Moreover IPIPGW does not have telephony Service Provider Interface (SPI) or telephony signaling stacks.So when a VOIP call arrives at IPIPGW, it matches inbound dial-peer and routes the call through outbound dial-peer. Each IP leg has its own RTP instance.
Media Controlling Modes:
We have 2 CLI to control media passing through IPIPGW.
Media flow-through:
The media is passed through the IPIPGW, the media will get terminated and re-originates with IPIPGW¿s IP address and port number, so here we cannot find the original gw¿s ip address. This is one of the security feature in IPIPGW.The RTP Packets will flow through IPIPGW
Media flow-around:
Here the media bypasses the IPIPGW. It just flows through the origination and destination gateway.
IPIPGW allows combination of connections like H-323 to SIP, H-323 to H323, SIP to SIP and SIP to H-323.
Deployment of Script:
In IPIPGW both dial-peers are VOIP dialpeer, so you can deploy the script eiether incoming or outgoing dial-peer.Based on the media controlling modes,the media passes through the gateway or bypass the gateway.
asg_com#conf t
asg_com(config)#voice service voip
asg_com(conf-voi-serv)#media flow-through or media flow -around

 

If we download the complete book length programming guide, the information tag leg_state and the status for that lg_xxx are missed.
But if we download chapter by chapter we can get those information.

Here is the link:

http://www.cisco.com/en/US/docs/ios/voice/tcl/developer/guide/tclivrv2_ch5.html


Leg State
The state of the call leg and the corresponding status code is reported in lg_xxx format:
Value for xxx Call Leg
000 LEG_INIT
001 LEG_INCOMING_FIRST
001 LEG_INCINIT
002 LEG_INCACKED
003 LEG_INCPROCEED
004 LEG_ALERTING
005 LEG_INCCONNECTED
006 LEG_INCDISCONNECTING
006 LEG_INCOMING_LAST
007 LEG_OUTGOING_FIRST
007 LEG_OUTINIT
008 LEG_OUTPROCEED
009 LEG_OUTRINGING
010 LEG_OUTCONNECTED
011 LEG_OUTDISCONNECTING
011 LEG_OUTGOING_LAST
012 LEG_DISCONNECTED

 



There is a perspective that TCL-IVR applications are not flexible enough for a newbie to play around. But in fact, there are lots of IVR applications that are coming inbuilt with Voice GW. You just need to copy/paste in a file editor and tweak it.

The recent hype is for BACD IVR applications. They are coming inbuilt with 12.4(15)T and later. Most of the enhancements are done with BACD AA application and that too coming inbuilt you can very easily play around with. Its easy to modify it to suite a big call centre/customer support kind of environment.

Just issue "show call application voice summary" to see all the inbuilt TCL applications. Happy IVR-ing :-)!!!

 

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