Hi
I've configured my ser server with dedicated Media proxy server. When I call
from IP to IP it works fine without any problem but when I try to dialout
using ser there is no audio. But if I'll enable STUN in my phone I'm able to
hear audio. my phones are behind Nat( Port rest con nat). Please guide me
how to setup my ser or mediaproxy so I don't have to use STUN server.
thanks
arun
I am getting a ACK loop after receiving a 404 from a remote GW. I am
starting the call using sipp for testing.
sipp -r 1 -l 1 -d 5000 -s 8989 -p 5061 -sn uac 10.14.18.22
The ngrep trace is showing Via and From set to 127.0.0.1 for the
communication between OpenSER and the sipp system. The ngrep output and
the routes from the openser.cfg are at
http://pastebin.ca/493807
I assume the Via/From being set to 127.0.0.1 may be the issue, which I
assume I could change from sipp, but I would also like OpenSER to catch
the loop.
Taylor
Hello all.
I've been experimenting with the new presence module in openser-1.2
and have a few questions which hopefully someone can help answer.
I'm using an SVN checkout of the 1.2 branch, r2237, and am testing
with Polycom SoundPoint IP phones, firmware 2.1.1.0052.
>From what I can tell, openser 1.2 does not fully support the xpidf
content type, it only supoorts the winfo content type. I've looked
through the code for the presence module extensively and it appears
that whenever a SUBSCRIBE comes in where the Event header is simply
"Event: presence" and the Accept header is "Accept:
application/xpidf+xml", the message is acceped and a 202 is sent in
reply but in any NOTIFY sent from openser as a result, there is no
message body and the message does not contain the XML information for
pidf.
Just as a test, I changed the event field in active_watchers to
"presence.winfo" and noticed that with this change, openser generates
a NOTIFY with a message body which contains watcherinfo. This, of
course, generates a 481 from the phone since it does not know about
the "Event: presence.winfo" event type.
Also as a test, I tried to use the pa module as I'd had success with
this module in the past since it _does_ support xpidf. In 1.2 it
appears that the database structure specified in the databse setup
scripts does not agree with the table structure being used in the
module. As a result, I could not get the pa module to work either.
Am I overlooking some small detail?
Please let me know if further information is needed in order to help
diagnose and resolve this issue.
Cheers,
-Curt
Hello, I'm new in Openser and now I'm trying to use a failure_route but it doesn't work and I don't know why. The version that I'm using is Openser 1.2. What I'm trying is to forward a call after fr_inv_timer to Asterisk but it doesn't work. This is my openser.cfg :
(...)
modparam("tm", "fr_inv_timer" , 8)
# ------------------------- request routing logic -------------------
# main routing logic
route{
(...)
if (uri==myself) {
if (method=="REGISTER") {
if (!www_authorize("172.16.51.150", "subscriber")) {
www_challenge("172.16.51.150", "0");
exit;
};
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
if(is_method("INVITE") && (uri=~"sip:2[0-9]+@")) {
t_on_failure("1");
}
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
failure_route[1] {
#redirect to Asterisk
rewritehostport("172.16.51.160:5060");
rewriteuser("2002");
}
Hi All,
I have followed a tutorial and set up Asterisk as a voice mail server.
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op…
It works fine when the UA is offline. Now, I want a call forwarded to the
Voice mail server when there is no answer from the UA after 60 seconds(UA is
registered on the openser).
What should I do? Below is my config (copy from the above link).
# requests for Media server
if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") {
route(3);
exit;
}
# mark transaction if user is in voicemail group
if(is_method("INVITE") && !has_totag()
&& is_user_in("Request-URI","voicemail"))
{
xdbg("user [$ru] has voicemail redirection enabled\n");
# backup R-URI
avp_write("$ruri", "i:10");
setflag(2);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
if(isflagset(2)) {
# route to Asterisk Media Server
prefix("1");
rewritehostport("10.10.10.11:5060");
route(1);
} else {
sl_send_reply("404", "Not Found");
exit;
}
};
# voicemail access
# - *98 - listen caller's voice messages, being prompted for pin
# - *981 - listen voice messages, being promted for mailbox and pin
# - *98XXXX - leave voice message to XXXX
#
route[3] {
# direct voicemail
if (uri =~ "sip:\*98@" ) {
rewriteuser("1");
xdbg("voicemail access\n");
} else if (uri =~ "sip:\*981@" ) {
strip(4);
rewriteuser("11");
} else if (uri =~ "sip:\*98.+@" ) {
strip(3);
prefix("1");
} else {
xlog("unknown media extension $rU\n");
sl_send_reply("404", "Unknown media service");
exit;
}
# route to Asterisk Media Server
rewritehostport("10.10.10.11:5060");
route(1);
}
failure_route[1] {
if (t_was_cancelled()) {
xdbg("transaction was cancelled by UAC\n");
return;
}
# restore initial uri
avp_pushto("$ruri", "i:10");
prefix("1");
# route to Asterisk Media Server
rewritehostport("10.10.10.11:5060");
resetflag(2);
route(1);
}
Thank you both very much for this hints, I will change the versions I
currently use immediately.
Regards
Sebastian
-----Original Message-----
From: Anca-Maria Vamanu [mailto:anca@voice-system.ro]
Sent: Thursday, May 17, 2007 5:55 PM
To: Ovidiu Sas
Cc: Schumann Sebastian; users(a)openser.org
Subject: Re: AW: [Users] OpenSER crashed with Segmentation Fault using
PresenceModule
I was just about to tell you this myself. Please take the svn version,
either 1.2 or the trunk.
Ovidiu Sas wrote:
> Hi Sebastian,
>
> You should use the 1.2 svn branch (or maybe the trunk). There are
> several fixes already present in the tree. There's no point testing
> against the original 1.2 tarball.
>
>
> Regards,
> Ovidiu Sas