Hi,
from the logs you see, that you proxy actually parses the "500 Try
later" reply, so it must have received it
You should check, from where you receive the 500-error; then you could
see, if you want to adapt your config.
Carsten
2010/8/24 Alejandro Mellado G. :
> Hi' Carsten,
>
> Of course
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
wrote:
> Let me make this a little clearer:
>
> 1) Your product is not particularly novel, interesting or new.
>
> 2) Commercial solicitations are not ap
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
Fixing former link:
http://widget.doddlephone.com/embed/webphone.jsp?sipserver=proxy.ideasip.com&username=deglk1&password=palindru&callto=1234567890&auto=yes&stun=stun.ideasip.com
">
Tel: +1 234 567 890
Sergio
On Tue, Aug 24, 2010 at 11:17 AM, Alex Balashov
wrote:
>
> You're an idiot.
>
>
> O
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:
http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.com&usern
This may be useful for calling Kamailio SIP server right from web pages /
browsers ( webphone/click2call)
For Instance, a web phone link to call:
http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.com&username=deglk1&password=4554&callto=1234567890&auto=yes&stun=stun.ideasip
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: ngoc217t
In that case, there is a network or transport-layer reachability issue
between the two clients.
On 08/24/2010 06:24 AM, truong ngoc THANH wrote:
Dear Alex Balashov,
thanks for helping
i try to disable force_rtp_proxy() in kamailio.cfg.but when i make
call, no stream transfer. the call can make
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]
Dear Alex Balashov,
thanks for helping
i try to disable force_rtp_proxy() in kamailio.cfg.but when i make call, no
stream transfer. the call can make but can not hear anything .
TRUONG NGOC THANH
Telecommunications Engineer
Tel: 0984 480 646
Y!M: ngoc217thanh
___
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 w
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:
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 (t_check_status("4
Hi,
I made an attempt at installing kamailio-3.0.3 in a box yesterday and ran into
a config file error on testing the installation.
0(3232): [cfg.y:] parse error in config file
/usr/local/etc/kamailio/kamailio.cfg line 24, column 6 : synthax error
0(3232): [cfg.y:] parse erro
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 pr
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
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