Hello,
I'm looking for some sort of gateway which can REGISTER itself at my SER
server, can respond to 407 messages and is directly connected to a pabx
(either using ISDN-2 or E1). So basically a SIP <--> ISDN-2 / E1 gateway
which acts as a user agent (or multiple).
The reason for this is that we don't want to replace the current pabx (which
only has ISDN-2 or E1 for external communication).
I've found some devices from Mediamatrix (http://www.mediamatrix.com), but I
would like to know if any more of you have come across the same problem and
which devices you use.
Some wonderfull ASCII art:
SIP |----| ISND-2 or E1 |----|
-------| |-----------------| |-- internal phone network
|----| |----|
Gateway? PABX
The reason I want the gateway to be able to authenticate itself is that the
gateway is located at another location(so the SIP in the picture above will
be going onto the internet), so I don't want people to be able to spoof the
IP of the gateway.
Kind Regards,
E. Versaevel
> the asterisk work with modem and card network or
> necesary with hardware?
> http://www.asterisk.org/index.php?menu=hardware
>
> the question is for costs?
> i am peru and not harware si very very cost.
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Hi folks,
I've been having litle troubles with SER.
When I offhook my BT100 and called another phone, I saw in
my log file that SER query the MySQL 3 or more times for the same
thing (in this case, about group membership) and the same for log messages.
I tried to flag the action but it doens't work.
I think that the real problem is that I'm missing some thing
in my configuration or I just didn't understand the call flow in SER.
any ideas?
I'm sending part of my ser.cfg, I hope it helps.
===================================
record_route();
if (method=="INVITE" && !isflagset(11)) {
~ log(1, "Voicemail is enable? ");
~ if (is_user_in("To", "voicemail")) {
~ log(1, "Yes\n");
~ setflag(8);
~ t_on_failure("1");
~ } else {
~ log(1, "No\n");
~ }
~ # native SIP destinations are handled using our USRLOC DB
~ if (!lookup("location")) {
~ #sl_send_reply("404", "Not Found");
~ if (uri=~"sip:[0-9]{4,6}@") {
~ if (!radius_proxy_authorize("")) {
~ proxy_challenge("", "0");
~ sl_send_reply("403", "That's not
your home");
~ break;
~ };
~ rewriteuser("901");
~ rewritehostport("x.y.z.w:5060");
~ log(1, "Not found\n");
~ t_relay();
~ break;
~ } else {
~ route(1);
~ break;
~ };
~ };
~ setflag(11);
};
=========================================
Thanks in advance.
- --
============================================
Rodrigo P. Telles <telles(a)devel-it.com.br>
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
============================================
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Hi everyone
Now i need to get some infomations about ser server running status.And i
want to know whether ser support SNMP,or whether there is a module like
SNMP.There is no snmp.so in my server.
What should i do to let ser server support SNMP?
_________________________________________________________________
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Hello,
I have a working config with ser cvs stable and I want to test new modules
from cvs unstable. If I try to use the same config file with the cvs
unstable I have errors with auth_db and db_url .
What has changed between ser cvs stable and ser cvs unstable ?
Nov 9 22:29:56 voip ser: set_mod_param_regex: parameter <use_rpid> not
found in module <auth_db>
Nov 9 22:29:56 voip ser: parse error (96,17-18): Can't set module parameter
Nov 9 22:29:56 voip ser: set_mod_param_regex: parameter <rpid_column> not
found in module <auth_db>
Nov 9 22:29:56 voip ser: parse error (97,18-19): Can't set module parameter
Nov 9 22:29:56 voip ser: set_mod_param_regex: parameter <db_url> not found
in module <uri>
Nov 9 22:29:56 voip ser: parse error (112,20-21): Can't set module
parameter
Nov 9 22:29:56 voip ser: parse error (197,54-55): unknown command, missing
loadmodule?
There is no issue when ser server runing with Public IP.But when
ser behind NAT ,sip client can't contact with ser Server.
What do I need to modify to make ser runing without any issue behind NAT ?
Can anyone give me a hint?Thanks
Hello,
I am student of final year Post Graduation. I have been given VOIP as my project for final exams. In the same respect I have to work and show the live demo of VOIP in our college Lab. I have one PC with Redhat 9.0 installed and downloaded Iptel SER 0.8.12 also have two hardware IP Phones. I have to demonstrate the communication between those phone using SER over our LAN. I will be highly thankfull to you if you can help me in setting up the same in details. Also need information abut how the networking needs to be done. Our Lab is behind NAT.
I have gone through the documentation proivided by the IPTEL, but as I am new to Linux and not able to understand how to configure the same. Please help me in this matter.
Thanks in advance.
Sarfaraz
==========================
Sarfaraz I. Chougule
PANVEL
Tel. 91-22-27490797
Office: 91-22-30380540
==========================
Hello,
I see that some ipphons have a forwarding service.
When this service is activated and the ipphone receive an invite he resends
a redirected message with the redirected number, so that the caller can make
an invite to this number.
But if this number is a PSTN number, then this call will be paid by the
caller and not by the user that had activated the redirection.
I want to know if it is possible to stop redirected replay or to change the
status of the response.
thanks
Laurent.
