thanks so much Tod. the problem was that pap2 extensions was bad configured.
and didnt register. I did with xlite. and work. but this is because you
drive me in the right direction. thanks so much. other problem is because I
want to provisining a polycom 601 but until now I dont know how. some said
On Dec 18, 2008, at 10:58 AM, Jose Colin wrote:
here the screen shot that you ask
On Thu, Dec 18, 2008 at 12:50 PM, Tod Fitch
wrote:
On Dec 18, 2008, at 10:31 AM, Jose Colin wrote:
HI. I have attached. a screnshot with the result of asterisk -
vv and the sip.conf and extensions.conf
On Dec 18, 2008, at 10:31 AM, Jose Colin wrote:
HI. I have attached. a screnshot with the result of asterisk -
vv and the sip.conf and extensions.conf
as I told the voip isp provider have same problem for incoming calls
"No route to destination" when attempting Dial(SIP/3624,120,r)
Three generic points to check.
In sip.conf, a registration line
In sip.conf, a context pointing to a context in extensions.conf
A context in extensions.conf to route the call
Examples: passwords removed to protect me!
In [general]
register=17476721176:x...@proxy01.sipphone.com
Also in sip.conf
;
On Dec 18, 2008, at 9:04 AM, Jose Colin wrote:
HI. thanks so much for the community. I have a very important
question that I already made but no answer and should be a easy
question for pros like you. so please if you have this question
please share it
the thing is that I install astlinu
HI. thanks so much for the community. I have a very important question that
I already made but no answer and should be a easy question for pros like
you. so please if you have this question please share it
the thing is that I install astlinux, it is said that the astlinux by
default is PBX-only s
