Hi,
Has anyone tried connecting the 'Cisco Unified Border Element' to OpenSER as
a H.323 /SIP signalling gateway to allow H.323 video calls to be placed over
OpenSER?
Can you see any problems trying to connect these two systems together?
Thanks
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Hi,
Has anyone tried connecting the 'Cisco Unified Border Element' to OpenSER as
a H.323 /SIP signalling gateway to allow H.323 video calls to be placed over
OpenSER?
Can you see any problems trying to connect these two systems together?
Thanks
--
View this message in context: http://www.nabble.com/H.323-signalling-gateway-for-OpenSER-tp23401428p23401…
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
Hello to all
Does someone knows of a Web application to allow users to add an account
in Kamailio?
SIREMIS is only for administrator use.
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespereira(a)startel.pt
Hi all:
Have running this script on Openser 1.2.3-notls version managing forwards
this way:
route[3] {
...
t_on_failure("1");
if avp_db_load("$ru", "*") {
if (is_avp_set("$avp(s:fwdbusy)/s")) {
setflag(23);
};
...
failure_route[1] {
...
if ((isflagset(23)) && (t_check_status("486"))) {
if (avp_pushto("$ru", "$avp(s:fwdbusy)")) {
t_reply("302","Moved Temporarily");
return;
};
};
...
User A calls user B and B has fwdbusy parameter set in user preferences; if
B is busy, proxy sends 302 message back to A with contact set to
$avp(s:fwdbusy) value. This setup usually works fine. However, sometimes the
proxy shows a strange behaviour. It starts to append an arbitrary contact to
the contact header in every forwarding it does. The contact appended has
nothing to do with A or B or the uri set in the avp. The same uri is
appended in all call forwardings performed in the system. If the proxy is
restarted, the problem disappears.
Example:
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.100.2.254:5060;branch=z9hG4bK51461DD5
From: <sip:30132@10.100.2.254 <sip%3A30132(a)10.100.2.254>>;tag=92DAB388-EE7
To: <sip:20050@domain.com <sip%3A20050(a)domain.com>
>;tag=880a5593aeb097bc75600b31d6e17107-78ac
Call-ID: C0C84A1C-33CE11DE-BEBEEAA9-C0D323DF(a)192.168.2.40
CSeq: 101 INVITE
Contact: sip:30050@domain.com <sip%3A30050(a)domain.com>,
<sip:030410@10.172.0.254:5060;transport=udp>;q=0
Server: OpenSER (1.2.3-notls (i386/linux))
Content-Length: 0
<sip:030410@10.172.0.254:5060;transport=udp>;q=0 is added in all forwardings
done the system.
Unfortunately I have no debug info cause it happens in production
environment. I couldn't reproduce the problem in test environment. Have you
got any idea on what could make it happen?
Thanks a lot:
Fran Lizaran
Hi,
Is it possible to get the XML response(i.e content other than text) in http_query function?
Thanks
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Hello ,
I use kamailio to have communication between sip client(bob) and MSN client
(alice) and i added msn address in bob budy liste(in x-lite)
so i want to have bidirectional communication without using purplemap !!
someone can help me or tel me witch "file.c" to modifie
thank you
--
------------------------------------------------------------
BAHA Rachid
Elève ingénieur en 3ème année à l'INPT, Rabat
GSM:(+33) 0619787609