Can they be installed on an HP UX platform. Considering they are written in
C, it shouldnt be a porblem, but has anyone tried that?
Thanks,
Rizwan Khan
Hello,
I'm figuring out the best approach to deploy a bridge between Websocket\Webrtc and SIP\rtp.
Can Kamailio (+mediaproxy-ng or something else) operate as a full Webrtc\SIP gateway (signaling, audio or video transcoding, ICE and so on)?
Some months ago I found the architecture described here http://www.kamailio.org/wiki/devel/rtcweb_breaker that proposes to introduce a new RTCWeb
Breaker.
Is it just a proposal or is Kamailio moving following this approach?
If Kamailio really requires a RTCWeb Breaker, what are the main issues against using Doubango webrtc2sip with Kamailio? Performance? Interoperability? License? ...
Thanks
Daniele
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is there a way to add debug statements in kamailio.cfg? I'd like to be able to dump some of the variables and also see which path the calls are taking.
thanks
Hello,
i receive many error on the failure route like this "
t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by
UAS: stored: 500, received: 487" . Do you know why ?
My version of Kamailio is the 4.1.2 and the code is this:
failure_route[MANAGE_FAILURE] {
if (t_is_canceled()) {
exit;
}
revert_uri();
if (!cr_next_domain("$avp(s:carrier)", "$avp(s:domain)",
"$rU", "$avp(s:host)", "$T_reply_code", "$avp(s:domain)")) {
exit;
}
if (!cr_route("$avp(s:carrier)", "$avp(s:domain)", "$rU",
"$rU", "call_id")) {
exit;
}
$avp(s:host)= $rd;
t_on_failure("MANAGE_FAILURE");
t_on_branch("MANAGE_BRANCH");
append_branch();
if (!t_relay()) {
exit;
}
}
Thanks to all
I am already have some practice to integrate Kamailio with Asterisk, when
all users creates and registers in Kamailio, and calls go to/from Asterisk
with static "host=kamailio_ip" settings for each user on Asterisk side.
I can't (don't know - how to) use in same way integration with FreeSWITCH.
Can't create in FS directory structure a user with "host=kamailio_ip", FS
require registration.
Maybe I can register user on Kamailio and send additional registration
request to FS with src ip changed to kamailio (lan ip)?
P.S. Reading FreeSWITCH 1.2 book in progress...
Hi,
After wasting most of the day trying to make mi_datagram over UDP socket
work. I eventually realize that it does asymmetric UDP communication, which
creates a lot of trouble for writing a useful MI script using PERL or
Python etc.
Anyhow, i go through the module code and was able to write a patch for
symmetric UDP communication using mi_datagram.
I have tested the patch using both Kamailio 4.1 and trunk branches and it
seems to compile and run flawlessly.
I am attaching here with this patch for Kamailio community. It would be
great if Kamailio developers would review and commit this patch to source
trunk.
Thank you.
I have a situation here,
when a dialog expires, it sends a BYE to both call legs and this does not
generate an entry on acc table.
however, when i use acc_db_request() on event_route[tm:local-request], i
get 2 entries on acc, 1 for the downstream, and 1 for the upstream.
is there a way of handling this so i can only get 1 BYE on acc?
Kelvin Chua
Hey community,
I'm trying to get a 2nd leg custom header value stored in my acc table.
Right now I'm appending the header with append_hf in a failure route and
trying to use:
modparam("acc","db_extra","custom=$hdr(Custom)")
A sip trace shows the header was properly added and xlog output shows the
proper $hdr(Custom) value just after I append the header, but the value
doesn't show in acc.
Does acc only read from the INVITE of the incoming leg?
Any suggestions on how to get this 2nd leg INVITE header in acc is greatly
appreciated.
Thanks in advance,
Ryan
Hi there.
I'm still trying to integrate Kamailio and freeswitch... where kamailio acts as a proxy and registrar ... and freeswitch provides conference calls and voicemail.
I have calls between two polycoms working and conference calls work.
But when I try to leave a voice message for a user by dialing 44+ext, the sip proxy never replies to the polycom.
I did a tcpdump and i can see the INVITE from the polycom to the sip proxy multiple times... but no response back. The phone eventually disconnects itself.
Here's what my config looks like: http://pastebin.com/wWgyVcxc
Just do a search for "route[FSDISPATCH]".
You will see how I check for the "44" prefix, and then send the call to a route called "FSVBOX".
Any suggestions would be appreciated.
1. route[FSVBOX] {
2. if(!($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.
I have a Kamailio server forwarding REGISTERs to an Asterisk box, similar
to the way Daniel shows here:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Can I force Kamailio to append rport=5060 to the topmost Via header, prior
to forwarding the REGISTER to Asterisk? I tried add_local_rport() but it
only included 'rport' and not 5060.
Thanks,
Marc