Hi all
I noticed that the desc_time_order parameter of the registrar module (in ser.cfg) doesn't seem to work as it should (according to the description in the registrar manual).
If I understood correctly, if desc_time_order = 1 (using modparam) the newest contact should be used among those of same priority (I'm using the append_branches parameter equal to 0).
Another doubt is whether it is possible or not to use append_branches=1 to do priority-driven forking (all contacts are receiving the request, no matter which is their priority; is it possible to limit the forking just to those of same priority (q)? or even something better? :-) )
thanks a lot
Jimes Vasco Milanez
Diretoria de Sistemas de Nova Geração
CPqD Telecom & IT Solutions
Tel.: +55 19 3705-4879
jimes(a)cpqd.com.br
www.cpqd.com.br
Hi,
I'm running ser-0.8.12 and radiusclient-0.3.2 on Redhat Linux 8. SER has
radius authentication and accounting enabled.
I have configured an ATA 186 to register to SER. When the authentication
request is forwarded to the radius server, the ATA's LoginID and UID fields
values are forwarded correctly but the password (PWD field) is blank.
I have tested runnning the "radexample" radiusclient test script and it
works perfectly: userid and password are received by radius server.
Any ideas on how to solve this problem ? Is there a way for SER to force
the ATAs UID value to be it's password value when sending authentication
request to radius server ?
Thanks.
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Hi,
Is it possible to change the connection information (c tag) in the SDP? My media gateway (SIP->PSTN) has local IP addresses. I made a NAT translation on the router to make it accesible to the public network. Softphones in the public network were to able to successfully registered on the SER proxy server. The problem is when i start to make a call. The OK message that is sent by the media gateway has an SDP whose connection information has a local IP since it doesnt know anything about the translation on the router. Although the signaling are all correct, the RTP ports are not able to successfully connect and as a result the parties are not able to hear each other. So I need to change the local IP on the SDP with the mapped public IP on the router before passing it on to the caller.
Thanks,
Ian
Hello,
can anyone explain to me the following error meesages
when I intend to start serweb by calling either
"user_interface" or "admin" index.php.
how to fix it.
=====================
Warning: Access denied for user: 'ser(a)mydomain.com'
(Using password: YES) in /var/www/phplib/db_mysql.inc
on line 73
Database error: pconnect(192.168.1.253, ser,
$Password) failed.
MySQL Error: ()
Session halted.
=======================
thank you/Linda
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Hi, can you send me a copy of the binary ngrep already patched for newline in SIP headers ?
The sip_router/utils/ngrep source is not compiling in my system.
Thanks
Ezequiel
Hi
I posted the below message a couple of days ago. I want all numbers
starting with 8 to go to a SIP PSTN gateway. The problem is that neither
forward() nor relay_to_udp() work through NAT. I want to be able to use
rtpproxy (or mediaProxy for that matter) for such connections. I see that
forward() and relay_to_udp() both bypass the nathelper rtpproxy. Anyone
knows the best way to ensure rtpproxy involvement in this scenario. If
rtpproxy is not involved, I cannot use the PSTN gateway from NAT'd
endpoints. There is gotta be a way.
/-------------------\
| SER (port 5060) + |
SIP UA <----- NAT ------> | rtpproxy |
| |
| SIP_PSTNGW |---->PSTN
| (* at port 7060) |
\___________________/
Previous post:
Hi
I am trying to send all my calls to a PSTN gateway (asterisk with digium
cards running on the same machine on 7060). I can use forward, but then I
cannot use the solution SER provides me by using NAT helper. Consider the
forwarding script:
      if (uri=~"^sip:[8].*@")
{
forward(209.7.34.58,7060);
    }
Here it works fine if I am calling from a public IP, but will not work if
I am calling from a NATed client since this stateless forwarding will
take SER out of the picture, and rtpproxy will not be used. I have also
tried using t_relay_to_udp, but is is doing the same, the below lines are
still not making SER to proxy the request.
if (uri=~"^sip:[8].*@")
{
t_relay_to_udp("209.7.34.58","7060");
break;
}
What would be the best way of doing it. I want SER to act as a proxy
(B2BUA?) between my caller and the PSTN asterisk server. I want to be able
to use nathelper for this scenario.
Regards
There is a new release of SER MediaProxy available. To update your
MediaProxy installation you need to perform the following steps:
1. cvs update on your unstable SER tree (for stable copy
modules/mediaproxy directory under your stable tree) and recompile
2. Download and install the latest server from
http://www.ag-projects.com/SER_Media_Proxy.html
Among various bugs fixes the performance has been optimized and the
billing features expanded. The live media sessions interface provides
web access to media streams from multiple servers including CodecType,
User Agent, Network traffic and Call status.
A snapshot can be seen at:
http://www.ag-projects.com/docs/MediaSessions.pdf
Regards,
Adrian
Hey can any one tell me from where can I get more info for b2bua and
from where can I download it and use it. I am a newbie in b2bua and I
already have my SER with Mysql running and now want b2bua on the ser.
Dipen K Gala
Fidelity Investments System Company - Telecom
Phone: (617) 563-2729
Email: Dipen.Gala(a)FMR.com
Has anyone installed or used B2BUA with ser. Pls share with me how to
start and from where can I download the AA, and do I have to code in
python in order to get it working the way I want???
Dipen K Gala
Fidelity Investments System Company - Telecom
Phone: (617) 563-2729
Email: Dipen.Gala(a)FMR.com
Hi all!
Some my sip clients are behind a nat on an internet public network that
"corrupt" UDP packect (wrong checksum) if is directed to port 5060.
Is possible to have a SER on port (for example) 5090 as a "proxy" for
the real ser on port 5060? How I have to configure ser.cfg for "fake"
ser on port 5090?
Thanks you,
Aldo.
