Call forwarding from PSTN line to PBX

[Udupikrishna091]

Hi,

I have a setup in my voice gateway where there is FXO card with 4 ports and there is a PSTN line connected to one of the ports. As of now i am making outgoing calls from my IP phone which goes through this pstn line. I have another line which is a Centrex connection [PBX] connected to another port, My requirement is if my basic pstn line is busy, my call should go through the other line, is this possible if the other line is a PBX line where i have to use '0' as the beginning number to get the line active.

Regards

Krishna

[Jonathan Schulenberg]

Sure. Just have a second dial peer with a higher preference value on it and the same destination-pattern value. You can use prefix 0 to get the trunk access code added.

[Udupikrishna091]

Hi Jonathan,

                         Thanks for the reply. the doubt i am getting is pretty lame, now for example if i connect the PBX line to a analog phone, i need to dial '0' then I will be getting the dial tone then i can dial a number!!!!, now if i have a prefix set in call manager, which would be discarded later. if 0 is my prefix and which would be discarded and I need one more zero to make the PBX line active, i cant send a single zero to the PBX through Call manager, if I dail 00XXXXXX, my first 0 will be discarded but when this goes to PBX it is dropped because it takes only zero, this is where exactly I am facing the issue. Is there any way that can make this work.

Regards

Krishna

[Jonathan Schulenberg]

Are you controlling the analog port through MGCP; or, using H323/SIP dial-peers between CUCM and the router?

If you're using MGCP: You can add the extra zero character by placing the two ports in separate route groups. The route list will allow you to do per-route group transformations. For the normal PSTN route group you add nothing. For the Centrex route group you prefix another zero.

If you're using H323/SIP: Just use 'prefix 0' on the outbound POTS dial-peer that is bound to the Centrex port.

[giorgiosalsiri]

If you are using MGCP and you add the new route group to your route list, all you need to match is your original route pattern (don't pre-pend anything).  From the orignal post, it sounds like you are not sending the leading 0 to your PBX, so it is not stripping it, and it is actually dialing 0 (when the call is delivered).  If you have it properly configured on your RP-RL-RG, you shouldn't have to pre-pend anything.