[SR-Users] Trying to dial 4000 into freeswitch's vmail system - Kam Proxy returning Status 403: Not allowed
mark li
limark67 at yahoo.com
Fri Mar 28 20:33:26 CET 2014
Hi there.
I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as a proxy and registrar ... and freeswitch provides conference calls and voicemail.
I have calls between two polycoms working and conference calls work.
But when I try to leave a voice message for a user by dialing 44+ext, the sip proxy never replies to the polycom.
I did a tcpdump and i can see the INVITE from the polycom to the sip proxy multiple times... but no response back. The phone eventually disconnects itself.
Here's what my config looks like: http://pastebin.com/wWgyVcxc
Just do a search for "route[FSDISPATCH]".
You will see how I check for the "44" prefix, and then send the call to a route called "FSVBOX".
Any suggestions would be appreciated.
1. route[FSVBOX] {
2. if(!($rU=~"^1[0-9][0-9]+$"))
3. return;
4. prefix("vb-");
5. route(FSRELAY);
6. }
7.
8. # Send to FreeSWITCH
9. route[FSRELAY] {
10. $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":"
11. + $sel(cfg_get.freeswitch.bindport);
12. route(RELAY);
13. exit;
14. }
15.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140328/91b7e82b/attachment.html>
More information about the sr-users
mailing list