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}/$1"/> </condition> </extension> And thanks for the tip, I will use variable instead. On Sat, Aug 22, 2009 at 4:35 PM, Michael Jerris <mike@jerris.com> wrote:
No, you don't get the full sip uri in the dialplan like that. You do have a whole bunch of variables of the parsed sip header you can use. Use the "info" application to see all the vars so you can see what you have to route the call on. Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
Hi:
I try to dial sip url from my softphone but seems like the sip address is being processed by sofia before it pass to the dialplan. The example here is :
*X-lite(softphone) dials -> 1009@4.2.2.2 (it's fake sip address, the purpose was just to test what's being passed to dialplan) sofia receives the invite and return with trying sofia pass the destination number to dailplan with "1009" (without the "sip:" in front and without the "@4.2.2.2" after it) * Please see pastebin for full log. http://pastebin.freeswitch.org/10089 ignore anything after line 80, because it's not my point, and the destination is a fake address.
I would like to know how do you actually pass a full sip url to the dialplan to do the regex match. Because from the default.xml dialplan, it comes with an example sip url dialing extension that match's *^sip:(.*)$ *. 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>
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-- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant