1 Dec
2014
1 Dec
'14
12:30 p.m.
I am not sure if its the issue on my client end as CSIPSImple is not sendin
the PRACK that is expected by the server or the rtp ports are not opened.
I guess the ports are opened as I can hear two way voice.
But the following issues remain:
1. I cannot receive a call on the SIP number
2. I dont get an indication on the caller (thru CSIPSimple) that a call has
started as in it doesnt start the call timer
3. The call disconnects with an error after some time ..INternal Server
Error.
Regards,
Rizwan Khan
On Mon, Dec 1, 2014 at 11:00 AM, Rizwan Khan <rizkhan(a)gmail.com> wrote:
Thanks for the help guys.
I used a stateless proxy and was able to register. Now I am able to make
calls as well but there is an issue.
I receive multiple ringing but not receiving 200 ok after that. I keep on
receiving the ringing messages. Although the call gets connected and I hear
two way voice as well.
Also I am call through the sip number, but its not receiving any calls.
Any ideas?
Rizwan Khan
On Thu, Nov 27, 2014 at 9:04 PM, Daniel Tryba <daniel(a)pocos.nl> wrote:
On Thursday 27 November 2014 08:29:51 Rizwan Khan
wrote:
We want to use Kamailio as the pass-thru proxy in
front of the SBC just
to
receive the request on a random port, and then
forward the request to
the
SBC internally being on the same private
network.
What would be the best way to do it?
If it is the best is up to the experts, but a way to do is to check
wheter the
source is the SBC or not. If the message is from the SBC just handle it
statefully (t_relay), if it isn't from the SBC set the requested domain
to the
ipaddr of the SBC (10.1.2.3) and process stateful. You could use the
received
port instead if external clients only communicate on the special port
($Rp!=6334)
kamailio.cfg:
#...
port=5060
port=6334
#...
request_route {
#...
if($si!="10.1.2.3")
{
$rd="10.1.2.3";
}
route(RELAY);
}
_______________________________________________
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
4 Dec
4 Dec
12:55 p.m.
New subject: Not getting the final response after ringing
Look at the sip traffic flowing though server (e.g., with ngrep on port
5060) and see where the packets are routed.
Cheers,
Daniel
On 01/12/14 11:30, Rizwan Khan wrote:
I am not sure if its the issue on my client end as
CSIPSImple is not
sendin the PRACK that is expected by the server or the rtp ports are
not opened.
I guess the ports are opened as I can hear two way voice.
But the following issues remain:
1. I cannot receive a call on the SIP number
2. I dont get an indication on the caller (thru CSIPSimple) that a
call has started as in it doesnt start the call timer
3. The call disconnects with an error after some time ..INternal
Server Error.
Regards,
Rizwan Khan
On Mon, Dec 1, 2014 at 11:00 AM, Rizwan Khan <rizkhan(a)gmail.com
<mailto:rizkhan@gmail.com>> wrote:
Thanks for the help guys.
I used a stateless proxy and was able to register. Now I am able
to make calls as well but there is an issue.
I receive multiple ringing but not receiving 200 ok after that. I
keep on receiving the ringing messages. Although the call gets
connected and I hear two way voice as well.
Also I am call through the sip number, but its not receiving any
calls.
Any ideas?
Rizwan Khan
On Thu, Nov 27, 2014 at 9:04 PM, Daniel Tryba <daniel(a)pocos.nl
<mailto:daniel@pocos.nl>> wrote:
On Thursday 27 November 2014 08:29:51 Rizwan Khan wrote:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
We want to use Kamailio as the pass-thru proxy in
front of
the SBC just to
receive the request on a random port, and then
forward the
request to the
SBC internally being on the same private
network.
What would be the best way to do it?
If it is the best is up to the experts, but a way to do is to
check wheter the
source is the SBC or not. If the message is from the SBC just
handle it
statefully (t_relay), if it isn't from the SBC set the
requested domain to the
ipaddr of the SBC (10.1.2.3) and process stateful. You could
use the received
port instead if external clients only communicate on the
special port
($Rp!=6334)
kamailio.cfg:
#...
port=5060
port=6334
#...
request_route {
#...
if($si!="10.1.2.3")
{
$rd="10.1.2.3";
}
route(RELAY);
}
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list
sr-users(a)lists.sip-router.org
<mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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
4:08 p.m.
New subject: Not getting the final response after ringing
I don't access the server. I only have access to the proxy. Server works
fine on the operators' broadband network. Have setup the proxy to make it
work on other ISP's network as well. So the proxy receives the traffic on a
unsual port which is not blocked and forwards it to 5060 port on the server
internally.
Rizwan Khan
On Thu, Dec 4, 2014 at 3:55 PM, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Look at the sip traffic flowing though server (e.g.,
with ngrep on port
5060) and see where the packets are routed.
Cheers,
Daniel
On 01/12/14 11:30, Rizwan Khan wrote:
I am not sure if its the issue on my client end as CSIPSImple is not
sendin the PRACK that is expected by the server or the rtp ports are not
opened.
I guess the ports are opened as I can hear two way voice.
But the following issues remain:
1. I cannot receive a call on the SIP number
2. I dont get an indication on the caller (thru CSIPSimple) that a call
has started as in it doesnt start the call timer
3. The call disconnects with an error after some time ..INternal Server
Error.
Regards,
Rizwan Khan
On Mon, Dec 1, 2014 at 11:00 AM, Rizwan Khan <rizkhan(a)gmail.com> wrote:
Thanks for the help guys.
I used a stateless proxy and was able to register. Now I am able to
make calls as well but there is an issue.
I receive multiple ringing but not receiving 200 ok after that. I keep
on receiving the ringing messages. Although the call gets connected and I
hear two way voice as well.
Also I am call through the sip number, but its not receiving any calls.
Any ideas?
Rizwan Khan
On Thu, Nov 27, 2014 at 9:04 PM, Daniel Tryba <daniel(a)pocos.nl> wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
_______________________________________________
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
On Thursday 27 November 2014 08:29:51 Rizwan Khan
wrote:
We want to use Kamailio as the pass-thru proxy in
front of the SBC
just to
receive the request on a random port, and then
forward the request to
the
SBC internally being on the same private
network.
What would be the best way to do it?
If it is the best is up to the experts, but a way to do is to check
wheter the
source is the SBC or not. If the message is from the SBC just handle it
statefully (t_relay), if it isn't from the SBC set the requested domain
to the
ipaddr of the SBC (10.1.2.3) and process stateful. You could use the
received
port instead if external clients only communicate on the special port
($Rp!=6334)
kamailio.cfg:
#...
port=5060
port=6334
#...
request_route {
#...
if($si!="10.1.2.3")
{
$rd="10.1.2.3";
}
route(RELAY);
}
_______________________________________________
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
3643
days inactive
3646
days old
2 comments
2 participants
participants (2)
-
Daniel-Constantin Mierla -
Rizwan Khan