Hello,
On 1/11/10 6:55 PM, Geoffrey Mina wrote:
this was from a regular route[] block. I ended up finding an old thread which pointed me in the right direction.
I am currently using this to retrieve the destination IP POST dispatcher lookup:
$var(destIP) = $(ru{s.select,1,:}{s.select,1,@});
It seems to be working well. Anything I should be aware of while using this technique?
actually ds_next_dst() populates the dst uri, which is accessible via $du. Make sure ds_append_branch parameter is set to 0: http://kamailio.org/docs/modules/3.0.x/modules_k/dispatcher.html#id2543896
Then you call append_branch() after doing your IP check.
If you do ds_next domain() then you get the new address in r-uri (therefore $ru).
To get access to dest uri domain, simply use: $dd.
Your expression with transformations will give you the ip along with protocol and uri parameters, if they are present in r-uri.
Cheers, Daniel
thanks.
On Mon, Jan 11, 2010 at 11:56 AM, Alex Balashov abalashov@evaristesys.com wrote:
Branch route?
On 01/11/2010 11:50 AM, Geoffrey Mina wrote:
What pseudo variable would i check after running ds_next_dst() to check the IP we are about to forward the INVITE to?
Basically I want to have a check for $si == [something] to ensure I am not about to send the INVITE to the same UI which just requested it.
While I have your attention, I may as well see if anyone has a better solution for what I am trying to accomplish:
I am doing this because asterisk can't handle a hairpin scenario and I need to generate two distinct calls in my network, so I figure i'll just forward the request to another server in the network to deal with this.
INVITE from PSTN --> Kamailio --> Asterisk#1 [Asterisk#1 needs to create a new session, but stay on-net] --> INVITE from Asterisk#1 to Kamailio --> [Dispatcher] --> INVITE from Kamailio to Asterisk#2
This would basically create 3 legs of the call, which is what I want
PSTN --> Asterisk#1 --> Asterisk#2
I would prefer if Asterisk#1 could just INVITE to LocalHost, but that doesn't seem to be allowed. Not sure if that's a limitation of SIP or Asterisk...
Thanks!
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