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[b_ball_henry at hotmai...]

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 -> 1009@4.2.2.2 (1009@4.2.2.2) (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it)
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the destination is a fake address.

I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's ^sip:(.*)$ . So I assume there must be a way of passing full sip url to the dialplan. Here is the example dialplan expecting sofia to pass it a full sip url:

 <!-- dial via SIP uri -->    <extension name="sip_uri">       <condition field="destination_number" expression="^sip:(.*)$">        <action application="bridge" data="sofia/${use_profile}/$1"/>       </condition>    </extension>
Thanks
--
Henry Huang

[mike at jerris.com]

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 wrote:

[red.rain.seven at gmai...]

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" expression="^sip:(.*)$">        <action application="bridge" data="sofia/${use_profile}/$1"/>       </condition>    </extension>

And thanks for the tip, I will use variable instead.


On Sat, Aug 22, 2009 at 4:35 PM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:

[jason at jasonjgw.net]

Henry Huang <red.rain.seven@gmail.com> wrote:

[red.rain.seven at gmai...]

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. but if I put "sip:" in the front of my dial string. The "sip:" gets trunkated by sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for regex matching. Therefore I say you can never reach the example sip uri extension because sofia will trunkate "sip:" .

Here is the excert from pastebin:
3.    INVITE sip:1009@4.2.2.2 SIP/2.0
(line 3 , the freeswitch has successfuly received my dialying to sip: 1009@4.2.2.2 (1009@4.2.2.2)) 73.  2009-08-20 16:37:28.982772 [INFO] mod_dialplan_xml.c:315 Processing 1001->1009 in context Global
74.  Dialplan: sofia/trunkgroup_1/1001@192.168.1.67 parsing [Global->number_1] continue=false

75.  Dialplan: sofia/trunkgroup_1/1001@192.168.1.67 Regex (FAIL) [number_1] destination_number(1009) =~ /^sip(.*)$/ break=on-false
(line 73~75, you can see that on line 73, sofia has trunkated the "sip: " & "@4.2.2.2" and only leave "1009" as the destination to pass to dialplan for regex match.)


On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason@jasonjgw.net (jason@jasonjgw.net)> wrote:

[mike at jerris.com]

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:

[brian at freeswitch.org]

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:

[red.rain.seven at gmai...]

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 <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

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:

[red.rain.seven at gmai...]

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 <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[red.rain.seven at gmai...]

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 name="sip_uri">       <condition field="destination_number" expression="^sip:(.*)$">        <action application="bridge" data="sofia/${use_profile}/$1"/>       </condition>    </extension>

On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian@freeswitch.org (brian@freeswitch.org)> wrote:

[brian at freeswitch.org]

You were told already this was used by mod_portaudio. So that you can pa call [url=sip:blah@domain.com]sip:blah@domain.com[/url] 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:

[msc at freeswitch.org]

On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven@gmail.com (red.rain.seven@gmail.com)> wrote:

[red.rain.seven at gmai...]

Michael:

Thank you for making it in "for dummies" format.
These are really nice tips I can use. thanks.

On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc@freeswitch.org (msc@freeswitch.org)> wrote:

[kevin at johnnyvoip.com]

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 Sat, Aug 22, 2009 at 1:35 PM, Henry Huang <red.rain.seven@gmail.com (red.rain.seven@gmail.com)> wrote: