Hi Claudius, thanks for your reply. More detail is provided below.
Richard
----- Original Message -----
From: "Claudius Fortis" <[EMAIL PROTECTED]>
To: <[email protected]>
Sent: Tuesday, October 09, 2007 9:40 PM
Subject: RE: [on-asterisk] Setup asterisk as a provider
Without much details on what you're trying to accomplish or what's failing,
I thought I'd try
- Are you thinking of allowing external users connect to your Asterisk box
via SIP or IAX protocol and make outbound calls?
--> If this is case, can local users register and call out
RK-> I bought a DID from a provider and their website allows me to map it
with SIP/IAX. It simply for incoming calls.
- Shouldn't your register line be in this format?
--> UserName:[EMAIL PROTECTED] and sometimes you might need to add
/Username at the end of your register line
RK--> I am currently using this format below but I do not get the CID of the
originating caller for the incoming calls.
extension1:[EMAIL PROTECTED]/extension2
Instead, I only get extension1 shown in the history.
- "What I got was a ringing tone but the actual destination did not ring"
--> Is the destination, username/extension 4162220000 ?
RK-> Destination can be anything ranging from ring group to individual
extension as I configure that with inbounf route in freePBX.
Hoping for some more details, have a great one!
Claudius
________________________________________
From: Richard (Rogers @ work) [EMAIL PROTECTED]
Sent: October 9, 2007 1:32 PM
To: [email protected]
Subject: [on-asterisk] Setup asterisk as a provider
Hi,
I would like to setup my asterisk to accept SIP or IAX incoming calls and
then use the inbound route to route calls to various destinations.
I setup a trunk with the following settings and use the following to
connect.
allow=ulaw
context=from-pstn
Secret=abcdef
Type=friend
Username=4162220000
[EMAIL PROTECTED] (no password)
What I got was a ringing tone but the actual destination did not ring.
I also have an inbounf route created for the 4162220000. I also went to the
general section to turn on the anonymous SIP incoming calls.
Any suggestions?
Thanks,
Richard
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