I am running openser behind the firewall.
and advertised_address is set correctly to the external ip address.
but record route seem to send the local/internal ip address.
help is appreciated.
Thx
Hi guys,
I am trying to route all calls from openser to our
commercial SIP Proxy server.
I modify PSTN.cfg file from examples folder and i
tried to call from one sip dialer but there is no any
request from openser to my SIP Proxy server. and the
dialer is registered and trying to call without any
error. I was try to look openser logs but there is not
error or warrning logs.
Same time i did capture from Ethereal and found "Time
Out" from openser to sip dialer.
Here is my pstn.cfg configuration::
loadmodule "/usr/local/lib/openser/modules/sl.so"
loadmodule "/usr/local/lib/openser/modules/tm.so"
loadmodule "/usr/local/lib/openser/modules/acc.so"
loadmodule "/usr/local/lib/openser/modules/rr.so"
loadmodule "/usr/local/lib/openser/modules/maxfwd.so"
loadmodule
"/usr/local/lib/openser/modules/registrar.so"
loadmodule "/usr/local/lib/openser/modules/usrloc.so"
# -- acc params --
modparam("acc", "log_level", 1)
# that is the flag for which we will account -- don't
forget to
# set the same one :-)
modparam("acc", "log_flag", 1 )
# ------------------------- request routing logic
-------------------
# main routing logic
route{
if (method=="REGISTER") { save("location"); return;
};
if (loose_route()) { t_relay(); return; };
if (method=="INVITE") { record_route(); };
if (!uri=~"sip:\+?[0-9]+@.*") {
if (method=="INVITE") {
sl_send_reply("403", "Call cannot be served
here");
} else {
# forward(uri:host, uri:port);
};
return;
}; #end of (!uri=~"sip:\+?[0-9]+@.*")
setflag(1);
if (method=="INVITE") {
if (!uri=~"sip:00[1-9][0-9]+@.*") {
sl_send_reply("403", "Forbidden");
return;
}; #End if (uri=~"sip:00[1-9][0-9]+@.*")
}; #End of (method=="INVITE")
rewritehostport("192.168.0.10:5060");
# forward the request now
if (!t_relay()) {
sl_reply_error();
return;
};
} #end of route
--------
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: abdulzu(a)hotmail.com
GoogleTalk: lateef.np(a)gmail.com
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
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Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice.
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-----Original Message-----
From: Kenny Chua <strain15(a)yahoo.com>
Date: Wednesday, Jun 28, 2006 10:56 pm
Subject: [Users] Using # for Sip 2 Sip calls
Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls?
I came up with something like this:
lookup("aliases");
if (uri=~"^sip:#[0-9]*@"){
xlog("Sip 2 SIP\n");
route(4);
route(1);
return;
};
Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you.
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--0-591390942-1151549737=:48905
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: 8bit
Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls?<br> <br> I came up with something like this: <br> lookup("aliases");<br> if (uri=~"^sip:#[0-9]*@"){<br> xlog("Sip 2 SIP\n");<br> route(4);<br> route(1);<br> return;<br> };<br> <br> Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you.<br> <p> 
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--0-591390942-1151549737=:48905--
[Users] Using # for Sip 2 Sip callsKenny Chua <strain15(a)yahoo.com>To: users(a)openser.org
Hi,
I'm wondering if the last version of SEMS is still compatible with OpenSER
1.0.X or more ?
Is someone using it there ? are the conferencing and voice mail feature
working well ?
Thanks,
Christophe
Can anyone point me in the right direction for resources that could help me
to improve call quality? I have everything up and running ok, but the
quality of the calls is a bit ropey at the moment. Any help here would be
much appreciated.
Thanks in advance,
Mike
Hi ,
A small issues, But i seems to be challenging issuses to all,
That...
UAC are behind the NAT.. are hunging up automatically..
That caller is not hung up, callee is hung up after 32 seconds.
And missed and rejecting call are working fine.
I think , ACK session time out and is not requesting to openser .
problem in openser.cfg with NAT or X-lite 3.0 softphones..
Problem in confiuration setting in softphones.
Or Problem with router Gateways ( NETGEAR) from outbound to inbound.
> Please any one give me suggestions or clues to resolve my issuess.
> > > > > > > > > >
> > > > > > > > >
Bye Happpen Weakend ,,,,,
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer && openSER)
Hyperion Technology
www.hyperion-tech.com
