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is its-CISCO. tcl supports SIP trunk

Sheraz Saeed Lodhi

Hi Folks, 

I am using cisco its-CISCO. tcl script for Auto attendant. I need to know if this supports SIP trunk between Telco and VG.


Thank you.


Manish Gogna
Cisco Employee
Cisco Employee

Hi Sheraz,

If you are referring to the topology like below, then it should work

++ Provider ---- SIP trunk ---- CME --- AA -- ++



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Yes Manish,

This is my topology then strange why did TAC say its not supported. Let me collect some more logs to investigate it further.

Did they say it is not supported or did they say it will not work?



He said this tcl script is not for SIP but the point is if this is the case then how come it is triggering the application and in application it doesn't find the destination number. I believe if it is not supported then it should not even trigger the app.

Hi Sheraz,

Have you solved the issue? I have the same topology and user can hear AA welcome audio but cannot dial the destination extension number (it seems that the script only collect first digit). Thanks.  

Hi Peter,

It didn't work for me then I have changed the Topology and used E1s as we had E1s as well and it worked for me.

What is the format of your AA welcome prompt as per my knowledge It only supports u-Law while in SIP case we were having A-Law and E1s are on u-law.  As per my knowledge Cisco AA tcl doesn't support transcoding therefore if we use transcoders in order to convert A-law to u-law it won't work.

If you are able to hear the welcome prompt as well as able to enter the digits then I believe this would be configuration issue.

Did you configure the voip dial-peer towards CCM to forward the call to operator or desired extension.

Hi Sheraz,

I will use the AA when router goes into SRST mode. I can hear the welcome audio well, and it will forward to operator if I don't enter any number. But if I want to enter 4 digits, it seems that only first digit will be collected.

I use G711ulaw  in dial-peer.

Hi Sheraz,

It works now. I've tried several days for this issue, but finally I found all the setting is okay except the test method. I always use my ip communicator to do the test, and I don't know why, it just can get the complete destination digit to send. I just use a phone and dial PSTN to the AA, it works. My goodness! Thanks for your help.


Hi Peter,

Good to know its working for you :). So are you using SIP trunk or ISDN E1 ?.

BTW for IP phone it should work as well because I have tested it with that one as well.

What is the type of your dial-peer for CUCM is it H323 or SIP.

I am using SIP trunk to SP (Verizon). I think maybe the issue for ip communicator is because I connect to the GW through VPN. My dial-peer is to the SIP SRST GW,and both sip and h323 are workable.

My topology is

Verizon SIP PSTN -- SIP trunk -- CUBE(SRST SIP VG) -- SIP Trunk -- CUCM

My scenario is:  When CUCM is unreachable, ip phones register to SRST SIP GW, and when outside people dial the pilot number, AA will answer and ask to enter ext. number or 0 to opetator.

Hi Peter,

Thanks for sharing it means it works for both H323 and SIP . As you said you are also using uLaw therefore I assume missing point from my side duting SIP was to not using or transcoding it to uLaw from aLaw.

Any way cheers you got your problem solved :)

maqsood ahmed
hi shiraz , would u post the relevant config from VG , of TCL script working for sip trunk, , mine it seems some problem,,
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