Hi'
I'm trying to forward the call to voicemail on sems when the time of
response is out. When the user isn't in location, the forward to
voicemail work very fine ( $rc = -1 ). But failure_route, doesn't work
and send message 500 Retry Later.
I'm using kamailio 3.0.2 and I'm probing with:
Hello All,
I'm using Kamailio as a Border controller for my VoIP Research project at my
school.
The problem I'm facing is Kamailio routes the traffic to the private network
where my asterisk machine is listening.
The asterisk machine responds to the Kamailio using the public network but
not the
Hi,
did you add a t_on_failure in your request route? Your route works
for busy subscribers i guess?
The problem is likely, that your sip-router box receives a 500 Retry
later from some endpoint (either the device or SEMS) and you have no
rule for handling 500-responses:
if
hi all,
i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so
how to configure in kamailio or which module make RTP stream direct from sip
client to another one ?
please suggest if anyone know.
thanks.
TRUONG NGOC THANH
Telecommunications Engineer
Tel: 0984 480 646
Y!M:
On 08/24/2010 05:41 AM, truong ngoc THANH wrote:
hi all,
i have using RTP proxy, and i see that RTP stream is handled by RTP
proxy. so how to configure in kamailio or which module make RTP stream
direct from sip client to another one ?
please suggest if anyone know.
On calls where you do not
Hi
Without rtpproxy or mediaproxy, the both SIP clients have to be reached from
Internet, or it has to have the public IP.
But in your case, I don't think you can have both client on Internet.
Tung
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org]
Hi Alex Balashov,
two clients is behind NAT, when i configure nathelper, the call make ok, but
RTP
proxy handle media stream, I want to make media stream go direct from sip
client
to another.
so is there any solve ?
TRUONG NGOC THANH
Telecommunications Engineer
Tel: 0984 480 646
Y!M:
This may be useful for calling Kamailio SIP server right from web pages /
browsers ( webphone/click2call)
For Instance, a web phone link to call:
a href=
You're an idiot.
On 08/24/2010 10:16 AM, Doddle WebPhone wrote:
This may be useful for calling Kamailio SIP server right from web pages
/ browsers ( webphone/click2call)
For Instance, a web phone link to call:
a
Fixing former link:
a href=
http://widget.doddlephone.com/embed/webphone.jsp?sipserver=proxy.ideasip.comusername=deglk1password=palindrucallto=1234567890auto=yesstun=stun.ideasip.com
Tel: +1 234 567 890
/a
Sergio
On Tue, Aug 24, 2010 at 11:17 AM, Alex Balashov
abalas...@evaristesys.comwrote:
Let me make this a little clearer:
1) Your product is not particularly novel, interesting or new.
2) Commercial solicitations are not appropriate for this list;
3) This product has absolutely nothing to do with Kamailio per se, and
your attempt to portray as though it does with the suggestion
Hi Alex,
Thanks for your feedback, I am sorry for any inconvenience I may have
caused.
Sergio
On Tue, Aug 24, 2010 at 11:31 AM, Alex Balashov
abalas...@evaristesys.comwrote:
Let me make this a little clearer:
1) Your product is not particularly novel, interesting or new.
2) Commercial
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