Hello
I have a question about uac module.
When i use uac_req_send function, from tag is made by this function automatically.
I wanna make a PRACK request which is in-dialog request, so i don't want to append new from-tag at that situation.
I think there some solution, but i can't find..
can you help me??
thank you
Hi,
I'm looking for a mechanism to unregister clients when the corresponding
TCP connection is closed, but had no success so far.
I've tried the handle_lost_tcp parameter of usrloc, but it didn't work:
the user was not unregistered, whereas the connection was seen as
closed by kamailio. Is it still a relevant option?
Otherwise, is there any mechanism that would trigger some kind of
event_route whenever a connection (TCP, TLS, WS, WSS) is lost or
closed?
Cheers,
--
Camille
Hi Dionlar,
can you elaborate the Memory issue a little bit more?
Can you also send me more information, on what OS and what
Kamailio-Version you are using? What configuration are you using?
If you have a memory issue, feel free to contact
sr-users(a)lists.sip-router.org, they should be able to help you as
well.
Probably no-one of our team (core-networks@) will be able to help you
further, as i am taking care of the implementation myself.
Thanks,
Carsten
2014-10-22 1:27 GMT+02:00 dionlar lang <dionlar.lang(a)gmail.com>:
> Hello dear!!
>
> I am Lang, undergraduate student research at the University Sheik Anta Diop
> of Dakar and I have a memory issue focusing on the interconnection between
> IMS and Webrtc.Je am writing to ask your proposals because I just read some
> of your articles about the subject.Is it you please help me.
>
> Thank you and I count on you!
--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Office +49 40 5247593-0
Fax +49 40 5247593-99
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
Hi
I didn't get any responses from anyone regarding my issue. Can someone please give me some clue what I can do? Or if more information is needed please let know so that I can provide.
Sent from my Samsung Galaxy smartphone.
-------- Original message --------
From: Kamrul Khan <dodul(a)live.com>
Date:10-21-2014 17:45 (GMT-06:00)
To: sr-users(a)lists.sip-router.org
Cc:
Subject: [SR-Users] One sided or no voice issue with websockets
Hi,
We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides. To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!!
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n")
.
.
.
route {
.
.
if (nat_uac_test("115")) {
if(nat_uac_test("64"))
force_rport();
}
if (method=="REGISTER") {
fix_nated_register();
add_rcv_param();
} else {
fix_nated_contact();
if(nat_uac_test("64")){
if (!add_contact_alias()) {
xlog("L_ERR", "Error aliasing contact <$ct>\n");
sl_send_reply("400", "Bad Request");
exit;
}
} else {
add_rcv_param();
}
}
}
.
.
onreply_route {
if (nat_uac_test(64)) {
add_contact_alias();
}
.
.
}
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
I read the article about asynchronous sqlops:
http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-4-asynchronous-sql.h…
I was wondering whether there are any plans of bringing that feature to the
acc module, too. We have seen it in the past that a blocking database
server brought our proxy to a halt because the acc module was blocking,
too. So this could help a lot if the acc entries were written
asynchronously.
Best Regards,
Sebastian
Hi,
I'm trying to re packet rtp streams based on Destination IP. I've
implemented it like below :
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
xlog("Destination [$du]...");
if ($du == "108.59.14.153")
rtpproxy_manage("coz20");
else
rtpproxy_manage("coz100");
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
rtpproxy_manage("coz100");
}
#!endif
return;
}
But its not working, and the log comes up like :
Oct 21 08:12:33 srv4617 /usr/local/sbin/kamailio[28346]: ERROR: <script>:
Destination...... <null>
Hi,
I've got a dialog database table with lots of old entries; some several
months old. I'd like to remove them, but if there is a method that keeps
things synchronous and checks beyond "DELETE FROM dialog WHERE .... "
I'd like to use it.
Any pointers?
--
Måns Nilsson primary/secondary/besserwisser/machina
MN-1334-RIPE +46 705 989668
I'm a GENIUS! I want to dispute sentence structure with SUSAN SONTAG!!
hi,
i want to forward call from VITELITY to sip TWILIO , by using config file
(attached one) call gets connected but end automatically after 33 seconds.
i could find some miss configurations in sip packet flow with routing ACK (
attached .pcap ) .
Please suggest a solution .
-regards
---------- Forwarded message ----------
From: Sadique Ali <spalikklpd(a)gmail.com>
Date: 21 October 2014 18:55
Subject: ACK not forwarding properly
To: sr-users-es(a)lists.sip-router.org
hi,
i want to forward call from VITELITY to sip TWILIO , by using config file
(attached one) call gets connected but end automatically after 33 seconds.
i could find some miss configurations in sip packet flow with routing ACK (
attached .pcap ) .
Please suggest a solution .
-regards
Hi,
We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides. To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!!
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n")
.
.
.
route {
.
.
if (nat_uac_test("115")) {
if(nat_uac_test("64"))
force_rport();
}
if (method=="REGISTER") {
fix_nated_register();
add_rcv_param();
} else {
fix_nated_contact();
if(nat_uac_test("64")){
if (!add_contact_alias()) {
xlog("L_ERR", "Error aliasing contact <$ct>\n");
sl_send_reply("400", "Bad Request");
exit;
}
} else {
add_rcv_param();
}
}
}
.
.
onreply_route {
if (nat_uac_test(64)) {
add_contact_alias();
}
.
.
}
