Your first question, I am not sure why ....but for this part i can explain abit
Also, on a side note, I have a context called [home] which each SIP
Phone is associated with. Do I need to specify each extension in
there?
SIP user can register as name as well . Doesnt means to have number. Example in sip.conf
[john]
type=friend
username=john
host=dynamic
context=incoming
secret=6769
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
Then in extensions.conf , you can have any number to ring this john sip user phone. Example :
exten =>9XXXXXXX,1,Dial(SIP/john) ; any number start with 9 end with 7 digit behinds. or you can also
exten => 9XXXXXXX,2,Hangup
exten => s,1,Dial(SIP/john) ; starting of the incoming call will ring John phone.
exten => s,2,Hangup
Hope my explaination is clear or fullfill your needs....thanks
On 6/17/06, John Klimek <
[EMAIL PROTECTED]> wrote:
Incoming calls from my Sipura 3000 don't seem to be correctly routing
to Asterisk (or something?)
Here is my Asterisk configuration for my incoming PSTN line:
Code:
[1000]
type=friend
host=dynamic
context=incoming
secret=6769
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
Inside of extensions.conf, I have this:
Code:
[incoming]
exten => s,1,Answer( )
exten => s,2,Background(enter-ext-of-person)
When I call my PSTN line, my Sipura 3000 seems to successfully answer
it because the line rings once, but then immediately switches to a
second dial tone. Shouldn't my incoming call be answered and then have
"enter-ext-of-person" played to them?
What could be causing this?
Also, on a side note, I have a context called [home] which each SIP
Phone is associated with. Do I need to specify each extension in
there?
For example:
exten => 50,1,Dial(SIP/50)
exten => 50,2,Hangup
exten => 21,1,Dial(SIP/21)
exten => 21,2,Hangup
Can't I just setup a default system where any two-digit number is
assumed to be an extension and it is automatically tried?
Thanks for any help!!
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