Hello everyone. I'm a student and I'm working on project which about SIP communication and PBXs with IP phones.
So, my first test results were OK with JITSI. But I think I need Freeswitch for working with IP phones. In a word, I need configure an integration Kamailio and Freeswitch.
First, I setup Debian 8.3 Jessie and installed Kamailio on this my local server, after that I installed another Debian 8.3 Jessie (from same image) as my second server and it was going to run Freeswitch on itself.
…
[View More]Then I tried to send my communication requests to my Freeswitch server. I made some tests, my Jitsi accounts which are set in Kamailio server were done. I've read Kamailio 3.2 and Freeswitch integration from the Kamailio wiki and I applied exactly the same thing but I did not work.
When I tried to reach another IP phone from my IP phone (I registered them) phone always say "NOT FOUND". I'm sure I did my configuration properly.
So, how can I figure it out these things? How can I understand, does Kamailio routes to Freeswitch to my phones?
Best regards.
Thanks.
[View Less]
Hello,
I have this scenario:
- subscriber registers using TLS
- call arrives from PSTN gateway (in UDP) to subscriber
- I relay the INVITE to subscriber but it doesn't answer the call
- the subscriber has callforward enabled
- we send the call to PSTN callforward destination using carrierroute
In the above, after calling cr_route I use this:
$du = "sip:" + $rd + ":" + $rp + ";transport=udp";
to force the call to PSTN gateway to go out as UDP because after trying to
call the …
[View More]subscriber, the transport changes and stays as TLS.
However, if instead of callforward the user has voicemail enabled, we use
dispatcher to send the call to a cluster of media servers.
So, after calling ds_select_dst()
I call
$du = "sip:" + $dd + ":" + $dp + ";transport=udp";
but this is unable to force the call to go as UDP but it still goes as in
TLS
In both cases I see this in the log files:
/usr/local/src/git/kamailio-4.3/kamailio[19277]: WARNING: <core>
[forward.c:231]: get_send_socket2(): protocol/port mismatch (forced
tls:XXX.XXX.XXX.XXX:5061, to udp:YYY.YYY.YYY.YYY:5060)
But it only works for carrierroute, not for dispatcher.
I am not sure if this is a bug in dispatcher.
But anyway, is there any other way I could try to force transport=UDP?
Regards,
Takeshi
[View Less]
I need to understand where from packets received. Now I use something like
If $si == "1.2.3.4" {
xlog("L_INFO","bla bla bla");
}
But I need to check source server not only by IP and PORT, but at Domain too
For example
if (some_pseudovariable=="pbx.server.com"){
xlog("L_INFO","bla bla bla");
}
I can use $fu for example because for some packets it includes domain name
of kamailio (i think details not important but this situations can be)
Does kamailio have some mechanisms to do that? I …
[View More]searched it at cookbook
but not found anything.
Thank you
[View Less]
