Hello All,
I'm looking for Ser Radius howto, or any other related documentation. For
some reason docs were removed from iptel website.
Regards
Alex Fler
Newbie to SER.
I am a little confused about how much I have to set up in databases to get
an incoming SIP call to call forward to a PSTN number through SER. I have
setup the username, attribute and value for my number, "callfwd",
sip:1234567890@gateway.com in usr_preferences and the username in the
subscriber file , added gateway ip to trusted, but call is immediately
dropped when it arrives at SER.
I have been through the getting started and have had success in modifying
the values in the scripts, but don't see what data must be added into MySQL.
Getting pointed in the right direction would be greatly appreciated
Thanx
murray
Hi;
I am using ser-0.10.99-dev35-pa-4.1 to talk with SIP client(UA) and XCAP server for handling subscription and publish.
Anyone can give me some comments for:
1. Difference between subscription handled by rls (handle_rls_subscription) and pa (handle_subscription).
2. Difference between table ser.subscriber and ser.rls_subscription.
Thanks
Smile
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---ST Electronics Group---
Hi;
I have a softphone (ie. Eyebeam) that talks with SER.
The question is whether Eyebeam can upload or download the contact list/buddy list file in xml format from XCAP server via RLS modules of SER.
Please advise.
Regards
Smile
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---ST Electronics Group---
Hi folks,
I have Asterisk as a PSTN termination GW, Openser will handle the user
registration.
Openser users has name format 7xxxxxxxx ( all digistes), which can call
each other peer to peer ( it is a basic openser function).
I want to forward the call ( 1xxxxxxxxxx or 011xxxxxxxxxxxxx ) to Asterisk
when openser user try to terminate US or International calls.
I tried to use the setting like this:
if(is_method("INVITE") && !has_totag() && uri=~"sip:011[0-9]+@.*") {
route(4);
exit;
};
route[4] {
strip(3);
# route to Asterisk Media Server
rewritehostport("222.222.222.79:5060");
if (!t_relay()) {
sl_reply_error();
};
exit;
}
OpenSER keep looking the 011xxxxxxxxx from location and got 408 always ,
it does not forward the call to Asterisk. I read a doc about how to
share acc and voice meail with Asterisk, will it have to use failure_route
to accomplish it ? I am new to Openser and dont know much about the routing
logic. Anybody could point me to the right way? A sample cfg will really
appreciated.
Steve
Hi,
I am cannot login to the initial Webpage for OpenserAdmin after starting
./script/server. I put in the default username:password of
openser:openserrw but I am not allowed access. Any suggestions anyone?
Thanks,
jswhel
To All
I answered it myself
if(uri=~"sip:002[0-9]*@")
Thanks
Rick
_____
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard C. Thompson
Sent: Tuesday, September 26, 2006 4:31 PM
To: serusers(a)iptel.org
Subject: [Serusers] digit matching
Hi All
I hope someone can point me in the right direction or just give me a hint.
I have the following route in ser.cfg and it works great. All I would like
to do is change it so it matches on any number that starts with 002
regardless of length not just 12 more digits or a fixed length.
if(uri=~"sip:(002............)@")
I tried
if(uri=~"sip:(002*)@")
but it didn't work
TIA
Rick
hello,
Trying to migrate from 1.0.0 to 1.1.0 I came across a problem that has
already been discussed in the list (of course if I had first searched
the archives I would have saved myself from a couple of days of
debbuging). Here are the two discussions:
http://openser.org/pipermail/users/2006-August/006246.htmlhttp://openser.org/pipermail/users/2006-August/006284.html
I am just sending this again because I think it looks like a bug to me.
In the avpops documentation for 1.0.0 the example says
avp_write("$ruri/username","$email");
In 1.1.0 it has changed to
avp_write("$ru/username","$avp(email)");
Neither of those two will produce the username in avp email with openser
1.1.0.
The same holds for
avp_write("$ru/domain","$avp(whatever)");
avp_write("$ru/domain","$avp(whatever)");
The solution is to use:
avp_write("$rU","$avp(whatever)");
avp_write("$rd","$avp(whatever)");
If the /username, /domain are not supposed to work, then shouldn't they
be removed from the document? And shouldn't they cause some syntax
error?
thanks a lot
George
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Is it possible to make SER to send Digest-Username(115) attribute defined in
RFC 4590 instead or in addition to Sip-Uri-Users(208) attribute?
С уважением, Колесников Андрей
Netris
<http://www.netris.ru> http://www.netris.ru
Hi,
I want connect my SER-Server to a public PSTN Gateway. The sample config
found in 'SER - Getting Started' doesn't work.
Someone out there who can help me?
Mit besten Gruessen
Ralf Eisele
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