[Asterisk-Users] Requirement for internal calls
Michi
mwuest at gev5.de
Thu Jun 23 23:04:55 MST 2005
Hi there,
I am new to Linux and Asterisk. I am not new to computer, network and
telecom stuff but only did 'Redmond-issues' and classic PBXs from
Siemens and Agfeo until now. So it took me some days to get linux (debian)
and asterisk up and running.
I made it to configure some SIP Extensions, with the help of AMP, and
I use X-Ten Lite SIP Softphones to do calls. The Login procedure works
fine. I have three internal phone numbers (200,201,202), but I can not
make internal calls from 200 to 201 or anything. When calling an internal
number X-Ten Lite tells me "404 Not Found"
I do not have configured any provider for the outside world. So I do
not have PSTN lines or SIP providers like sipgate configured. Now I have
these questions:
- Do I need to have these outside world "trunks" configured to tell
asterisk that when I dial "0" I want to dial outside, and when not,
that I want to dial an internal number?
- Is it the G729 codec and license problem?
- Any other hints?
For SIP I also have these questions:
- Is an sipgate account busy when I am already taking or making an call?
Or can someone else use a "2nd line" of an sipgate account?
- Is it possible to have dial-through numbers through an sip-account,
meaning my sip-number is for example 123456, then the person with the
internal extension 200 will have as an direct call through number
123456-200 ??
Thanks in advance,
bye,
Michael.
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