[SR-Users] alias problem

[Anca Vamanu]

Hi Mihaylov,

If your Asterisk servers add a Record-Route header to the initial 
Invite, for in-dialog requests ( ACK, BYE) you should use *loose_route() 
*function to do the routing. This will make sure the requests go the 
same path as the initial Invite. It is not a good practice to manually 
route these requests.

Regards,
Anca


On 01/29/2012 11:10 PM, Stoyan Mihaylov wrote:
> My whole configuration is:
> [Sip clients] < = > Kamailio 3.2 <=> Asterisk servers (behind Kamailio)
> Asterisk servers have only local IP addresses, and I use t_relay 
> instead of forward.
> Kamailio runs on same server as rtpproxy.
> Everything is fine if clients connect to Kamailio with its IP address 
> - global, or if they are behind Kamailio with local address.
> When clients connect to Kamailio using sip.ourcompany.com 
> <http://sip.ourcompany.com>, then call (video also) is OK, but ACK and 
> BYE do not work.
> BYE receives not here (404), and ACK die somewhere.
> I forward BYE and ACK in case when src_ip==$td to Asterisk server.
>
> If one of clients use IP - then calls initiated from it are OK 
> (BYE/ACK - are going correctly - to Asterisk and to other client 
> also). But calls from other client have problems with BYE and ACK.
>
> To use sip.ourcompany.com <http://sip.ourcompany.com> - I put:
> alias=sip.ourcompany.com <http://sip.ourcompany.com>
>
>
> route[ACKBYE] {
> #!ifdef WITH_PSTN
> if (is_method("BYE|ACK"))
> {
> xlog("L_ALERT","AB $rm $sht(forw=>$ft) $td");
> if(src_ip==$td){
> #I have to rewrite du - messages loop in Kamailio, I store 
> in $sht(forw=>$ft) $du which I use during INVITE.
> $du=$sht(forw=>$ft);
> route(RELAY);
> exit;
> }
> xlog("L_ALERT","ACK,Bye Not me");
> }
> #!endif
> return;
> }
>

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