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.
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.
Hi,
I am not sure, if its a kamailio issue or something else, but our Kamailio 4.2 server crashes everyday. Throwing out these logs: http://pastebin.com/z1MWdk7q. We have to restart the server to get this fixed. Please help us understand what is causing the issue.
Thanks and Regards,
Sharon Nathaniel
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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.
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.
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 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
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 searched it at cookbook
but not found anything.
Thank you
Dear All
I m trying to setup kamailio and asterisk in load balancing with a2billing
. Currently all of my VMs, one Kamailio and two asterisks are on same
subnet . I have started the RTPproxy like below
./rtpproxy -s udp:127.0.0.1:7722 -l X.X.X.153 -m 10000 -M 50000 -u root
root -F -d INFO LOG_LOCAL0
My question is if all the VMs are on same subnet with same gateway what
should be written in the private IP X.X.X.153/<private - ip>
Secondly i m authenticating and registering the SIP on kamailio using the
A2B DB . all the dialplan for a2b is being run on asterisk . Now if i want
to call SIP peer to Peer like in case of followme case ,
How should i route the calls in Kamailio ? will it be using usr loc module?
if so any help will be appreciated
--
Regards
Ahmed Rehan
Hi all,
we’re just playing around with an IMS setup bases on kamailio. Therefore the kamailio is used as P-,I- and S-CSCF.
We where able to register two clients through all components.
As we wan to start a call session, the P-CSCF answers with „403 - Forbidden. You must register with an S-CSCF“…
I found this snippet in kamailio.cfg:
if (!pcscf_is_registered("location")) { send_reply("403","Forbidden - You must register first with a S-CSCF“); break; }
Can one tell me what exactly the kamailio is checking there? As I figured out it is looking in database location table.
The table contains the registered users registered towards the IMS.
Another „problem“ we faced with is the rtpproxy. As the ngcp-mediaproxy-ng is no longer available and replaced with rtpproxy we’re trying to use it. But kamailio said that the proxy isn’t answering the way expected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid response
Any advise would be nice.
Thank you
Fred
Fred,
I think you would need to use rtpengine instead of rtpproxy.
https://github.com/sipwise/rtpengine
-Aaron
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Fred Schulz
Sent: Thursday, January 28, 2016 1:24 PM
To: sr-users(a)lists.sip-router.org
Subject: [SR-Users] P-CSCF
Hi all,
we’re just playing around with an IMS setup bases on kamailio. Therefore the kamailio is used as P-,I- and S-CSCF.
We where able to register two clients through all components.
As we wan to start a call session, the P-CSCF answers with „403 - Forbidden. You must register with an S-CSCF“…
I found this snippet in kamailio.cfg:
if (!pcscf_is_registered("location"))
{ send_reply("403","Forbidden
- You must register first with a S-CSCF“);
break;
}
Can one tell me what exactly the kamailio is checking there? As I figured out it is looking in database location table.
The table contains the registered users registered towards the IMS.
Another „problem“ we faced with is the rtpproxy. As the ngcp-mediaproxy-ng is no longer available and replaced with rtpproxy we’re trying to use it. But kamailio said that the proxy isn’t answering the way expected.
ERROR: rtpengine [rtpengine.c:1622]: rtpp_test(): proxy responded with invalid response
Any advise would be nice.
Thank you
Fred
