I am trying to route to a BroadVoice PSTN gateway using the UAC. In the
failure_route the uac_auth command fails with the following errors:
ERROR:uac:parse_authenticate_body: parse error in <DIGEST
realm="BroadWorks",algorithm=MD5,nonce="1133904594381"> around 0
ERROR:uac:uac_auth: failed to parse auth hdr body
code:
failure_route[3] {
if( t_check_status("401|407") ) {
if( uac_auth() ) {
append_branch();
route(1);
}
}
}
Thanks,
Hi,
I am using STUN and that works well most of the time.
I do have a problem with UA's who sit behind a symmetric firewall.
( a Microsoft server with ISA 2004 to be exact ). I understand this
is a well known problem. What practical solutions can I use in
combination with openser ?
Thanks for any help
Best regards
Paul van Schagen
Hello
Im managing a WAN with a lot of Universities. Some of them already
installed a VoIP solution based on SER (to manage SIP clients) and
Asterisk (for services and PSTN GW). The DNS routing provided by SER is
working perfectly, but we want to start routing all calls thru IP
transparently.
We want our legacy PBXs (that are connected to Asterisk) to forward all
calls to IP. The idea is to forward all calls to a central VoIP server,
that has all the numbers that already are VoIP enabled, and then:
- if the called number is VoIP enabled, he routes the call to that Univ.
VoIP server
- if the called number isnt in the list, the call goes back to the PBX
and a PSTN call is dialed
This way, ppl starts using the VoIP infrastructure, without even knowing
what VoIP means, and the telecom bill starts decreasing.
I know thats a statical and hierarchical structure and we dont want
that, but is a good solution for this migration phase, where a lot of
places are still using TDM systems.
Now, the top of the hierarchy should be an Asterisk or SER? I dont know
which of the systems is the best choice for the job. Does someone has an
idea of what should we use?
Thanks
Joao Pereira
www.fccn.pt
hi,
i have installed the SER in a REd hat linux machine.
i want to write a test stub in TCL , and the stub will throw SIP messages to
SER and will receive the same from SER.
any one , please guide me in this regard.
Thanks in advance
Rgds,
John
The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in
Hi,
I've been playing around with TM callbacks and the TM API trying to
build a module which mantains dialog info in order to permit session
termination. For now I managed (in a very experimental way) to have a
function exposed to the script which initiates (appropriately built
with dialog info) BYE transactions to UAs and effectively terminates
the call. Later this could be done via FIFO or unix socket command
taking some kind of "dialog identifier" parameter...
But sending BYEs to terminate calls only works if the dialog is
already in the CONFIRMED state.
For early dialogs, CANCELs should be used. But in the TM API there is
no cancel function. Is there any other way of canceling a request on
behalf of an UA from within an (open)ser module? Or would the TM API
need to be extended?
Thanks in advance for any insights.
JF
How can we change the database, database user name and database password that the openserctl utility tries to connect with?
You can set these in openser.cfg with modparam, but if you can't also set them for openserctl, what's the point?
Thanks.
Douglas Garstang
Hello,
I'm trying to setup a hunt group using the lcr module and different q
values. When I try to add a user location with a different q value
using the FIFO commands, the only q values that are accepted is 1.00 or
lower, such as 0.99. However, these all seem to qualify as the same
value as all of my locations are still contacted at the same time. Does
anyone have this working, and how did you add the user locations in
regards to the q value?
Thanks,
Brian
What is wrong with my PHP code, I have no idea. Could you please check
it and let me know what's wrong. Im using SER 0.9.3, but ser_fifo
documentation is not available, so i used one from ser 0.8.12 (admin
guide). Im sure there's a little error, but I can figure out which
one. Btw, below the code is the output from message log
<?
/* config values */
$domain="sip.mydomain.org";
$user="webme";
$web_aor="sip:".$user."@".$domain;
$expires=3600;
$type="student";
$fifo="/tmp/ser_fifo";
/* open reply fifo */
$replyfilename="webfifo_".rand();
$replypath="/tmp/".$replyfilename;
echo "Initiating your request...<p>";
/* open fifo now */
$fifo_handle=fopen( $fifo, "w" );
if (!$fifo_handle) {
exit ("Sorry -- cannot open fifo: ".$fifo);
}
/* construct FIFO command */
$fifo_cmd=":ul_add:".$replyfilename."\n".
"location\n". //table
$web_aor."\n". //aor
"sip:".$user."@".$_SERVER['REMOTE_ADDR']."\n". //contact
$expires."\n". //expires
"0.00\n". //priority
"0\n\n";
/* create fifo for replies */
system("mkfifo -m 666 ".$replypath );
/* write fifo command */
if (fwrite( $fifo_handle, $fifo_cmd)==-1) {
unlink($replypath);
fclose($fifo_handle);
exit("Sorry -- fifo writing error");
}
fclose($fifo_handle);
/* read output now */
if (readfile( $replypath )==-1) {
unlink($replypath);
exit("Sorry -- fifo reading error");
}
unlink($replypath);
echo "<p>You are now signed in<p>";
?>
///MESSAGE LOG
Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: read_line:
request line too long
Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: ul_add: flags expected
Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server:
command must begin with :: location
Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server:
command must begin with :: sip:webme@203.159.32.50;type=student
Dec 6 16:35:24 sip /usr/local/sbin/ser[206]: ERROR: fifo_server:
command must begin with :: 0.00
Dec 6 16:35:54 sip /usr/local/sbin/ser[206]: ERROR: ul_add: flags expected