Auto Attendant on SRST/CME gateway without CUE
August 4th, 2008At times it is not necessary for voicemail boxes or advaced auto-attendant features on a remote gateway that is serving a small office. This is especially true for remote non DID (ie FXO trunked) SRST gateways that are only active a small amount of time. Instead of adding a AIM-CUE or NM-CUE voicemail module in the gateway, basic auto attendant features can be provided right in IOS.
This is through the use of tcl scripting and the creation of applications. Cisco provides the tcl script for this purpose on their website at the SRST downloads page.
The basics for the script include the following components:
- tcl script – This is copied to the flash of your IOS gateway. For this example I am using the file from Cisco.com “its-CISCO.2.0.1.0.tcl”
- au files – The tcl script os packed with the audio files the script plays back to the caller. These are generic, but can be re-recorded to customize the feedback.
- application – an application will need to be created that calls the tcl script and passes the required variables.
- dial-peer – a dial-peer needs to be created to call the application.
- Enable the routing of calls to this extension.
- Additionally you will need to know what extension you want to assign to this auto attendant as well as teh extension of the operator for calls that are not matched.
Note that this is a VERY simple auto attendant. It simply answers, plays aa greeting, then asks the caller to enter the extension of a user. If the caller enters and invalid extension, it reprompts up to 3 times. If the caller enters 0, it transfers to the operator extension.
First make sure you have copied all the files from the download to you flash. You can place this in a subdirectory or the root directory of the flash. Just make sure you know where it is located. This will be important when we create the IOS application next.
Second, create the IOS application. For this example, I have coped the files into the root of the flash and I am using the extension 4999 for the autoattendant. Additionally, I am specifing the operator at extension 4001. The format for this is as follows:
!
application
service nicknetaa flash://its-CISCO.2.0.1.0.tcl
param operator 4001
paramspace english language en
paramspace english index 0
paramspace english location flash://
paramspace english prefix en
param aa-pilot 4999
!
Finnally, create the dial-peer to call the application. Again, I’m using 4999 as the extension for the auto attendant.
!
dial-peer voice 4999 voip
service nicknetaa
destination-pattern 4999
session target ipv4:1.1.1.1
incoming called-number 4999
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
Note that I am refercing the local loopback ip address (1.1.1.1) for the dial-peer destination. It is important to use a local interface ip that will not be “down” in the event of a link failure.
If you have a phone that is registered to the gateway, you should be able to dial it directly.
At this point, the application is configured and you simply need to route calls to it. This can be done in many differnt ways. If you are getting DID inbound from the provider, simply create the dial-peer and application as the extension of the main company number. Otherwise, the default-destination command under call-manager-fallback can be used. Another method if you have FXO trunking is to “connection plar” each of the voice FXO ports to the AA extension.
Tags: HOWTO
January 14th, 2011 at 9:00 am
I realy appreciate you explanation.
Very usefull.
Thanks you.
Daniel
January 10th, 2012 at 11:50 pm
Very informative article, and great blog. Thanks!
January 31st, 2012 at 7:09 am
Thanks,
Really useful article. I setup the above and working fine. I have few issue with the setup.
1.) When i use its-cisco-2.0.3.0.tcl it works fine but only two option (dial by extn & operator). I also require to dial by pressing individual digits eg: press 1 for sales, 2 for marketing etc
2.) Sometimes when an external line attempts to connect through the system, but the external POTS phone disconnects before the IP phone user picks up – the IP phone will ring FOREVER – until the IP phone is taken off-hook.
Like in this example… A user in POTS calls the business number after hours. Nobody inside the network picks up the phone. But it continues to ring, continually – until users come to work in the morning and off-hook their phones.
Is there any way to limit the number of rings on inbound phone calls from POTS, prior to simply disconnecting them?
Pls provide me a solution…
raffi_udeen@yahoo.com
Thanks & Regards…
Raffi