Outgoing CallerID on SIP ? Also, call termination

Asterisk Asterisk Support
1

Hi,

I have spent COUNTLESS hours in the last few weeks trying to get an asterisk box up and running.

So far, I have the following working:

PAP2 connected, both lines active (seperate extensions)

X100P connected, works, but outgoing audio flat out SUCKS or is not even intelligiable (doesnt matter weather the call is incomeing or outgoing). INcoming audio is good, though. I have heard this is caused because my handset (pap2) is using g729. I am going to try ulaww (g711) tonight.

Cisco 7960 connected, has dial tone, WILL NOT receive or place calls – trys but then gets fast busy

all incoming calls go to a ring group (so all the phones ring) then to a specific mail box.

Ok, here are my questions

  1. When I place an OUTGOING call over my SIP trunk, the outgoing caller ID (display to the person I am calling) shows “unavailable”.

How do I fix that?

  1. How do I make incoming calls from the SIP trunk terminate to a DIFFERENT ring group then the calls from my land line (x100p wildcard)?

thanks for any help ya’ll can gice d:o)

—Ray

2

sorry, my closing line said:

Thanks for any help you can gice d:o)

It should have said:

Thanks for any help you can give d:o)

Big fingers, small keys :o)