I have cisco 7940's and 7960's with a 2821 router and CUCM 7.1.2. When the phones go into SRST, I can dial out and caller ID verifies the line I am calling out is correct and the phone has the correct line appearence, but dialing in does not work.
You want to see how many digits are coming in from your provider, and reference that to the length of your DNs on CM. The phones register
to SRST with the extension they are configured with from CM.
So if your provider sends 1001, and you have 5553451001 configured on CM for the extension, the call will fail.
Also, if this is MGCP, make sure you have it configured to switch to the local call API when MGCPapp is down:
service alternate default
If you want to collect 'debug voip ccapi inout' and 'debug isdn q931' along with 'sh call-manager-fallback dial-peer' for an inbound SRST call, then it should be easy to figure out.
The previous poster is right, that you can use dialplan pattern to add digits ontot he front if the provider is presenting less digits than you have your extensions configured for. You can use a translation profile to do it, too. Or use num-exp, but that's a old & sloppy way.
I am going to try the debug, but in the mean time, I wanted to post what I had on the router to make sure that part is setup correctly. The phone numbers are fake, I didn't want to display the real ones
I was able to do "debug voip ccapi inout" but "debug isdn q931" had a problem with the isdn part. Anyway the "debug voip ccapi inout" was very useful, as it showed the call was erroring out, but I can't figure out where it states what the provider is sending
I take the calls aren't coming in a PRI if that debug wasn't available work. What type of PSTN circuit are you using? If it is an analog FXO circut, no digits are sent, and you need to issue 'connection plar ' under the voice-port.
Otherwise, CCAPI should show the digits being sent by the provider, if it is something like T1-CAS.
Since FXO ports can't receive a dialed number, you need to configure 'connection plar ' on the voice-port, and force the call to route to that specific extension. If you look at your CM config, you'd see that these ports have an 'attendant DN' configuration going to a default extension. Ideally, you'd send it to an extension on the receptionist's phone. The other option is to send it to an AA with dial by extension, but you'd need an AA local to the SRST router to make that work.
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