Hi:
I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:
*X-lite(softphone) dials - 1...@4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia
No, you don't get the full sip uri in the dialplan like that. You do
have a whole bunch of variables of the parsed sip header you can use.
Use the info application to see all the vars so you can see what you
have to route the call on.
Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang
It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:
!-- dial via SIP uri --
extension name=sip_uri
condition field=destination_number
Henry Huang red.rain.se...@gmail.com wrote:
It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:
It is entirely possible to reach this extension, but
Jason:
I fully understand how the regex works in the dialplan. If you look closely
in my original email and check out the pastebin. You will see that sofia
does not pass the sip: to dialplan. I can do any combination of letters
that dials from my softphone, and it will pass them to the dialplan.
a call coming from sofia would never hit that in the dialplan. That
extension is useful for dialing a sip url from mod_portaudio.
Mike
On Aug 22, 2009, at 10:09 AM, Henry Huang wrote:
Jason:
I fully understand how the regex works in the dialplan. If you look
closely in my original email
Remember the dialplan is agnostic... it has no clue about SIP, IAX,
Jingle, H323... it routes... you have various other variables you can
condition on also... route on destination_number and you'll be fine.
/b
On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
I fully understand how the regex
Brian:
but why can't I pass sip: to dialplan? seems like it's being truncated by
sofia..
Can you confirm that?
On Sat, Aug 22, 2009 at 10:30 PM, Brian West br...@freeswitch.org wrote:
Remember the dialplan is agnostic... it has no clue about SIP, IAX,
Jingle, H323... it routes... you have
Because the dial plan is technology agnostic... you have been told
more than once it won't pass it to the dialplan from mod_sofia...
/b
On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
Brian:
but why can't I pass sip: to dialplan? seems like it's being
truncated by sofia..
Can you
Brian:
Sorry, it's my English. I didn't understand what you meant by agnostic
back there. Now I know.
Thank you.
On Sat, Aug 22, 2009 at 10:59 PM, Brian West br...@freeswitch.org wrote:
Because the dial plan is technology agnostic... you have been told
more than once it won't pass it to the
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample sip uri extension in the default.xml dialplan. because
no one can reach the dialplan with prefix sip: because sofia is going to
remove that prefix.
!-- dial via SIP uri --
extension
You were told already this was used by mod_portaudio. So that you can
pa call sip:b...@domain.com which portaudio passes the exact string
you dial with pa call to the dialplan.
/b
On Aug 22, 2009, at 10:07 AM, Henry Huang wrote:
Brian:
Oh, and again, if it's not passing it to the
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang red.rain.se...@gmail.comwrote:
Brian:
Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample sip uri extension in the default.xml dialplan. because
no one can reach the dialplan with prefix sip: because sofia
Michael:
Thank you for making it in for dummies format. :P
These are really nice tips I can use. thanks.
On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins m...@freeswitch.orgwrote:
On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang red.rain.se...@gmail.comwrote:
Brian:
Oh, and again, if it's
X-lite I believe handles the sip: by itself sometime and therefore will try
and place a call to the sip address directly from x-lite without touching
FreeSWITCH. Be aware of this while testing and watch for this behavior
because it might throw off your expectations.
Regards,
Kevin Green
On
15 matches
Mail list logo
