Hi,
I am searching for cpled. I searched iptel.org, google, and the mailing
list. The only thing I could find was this page
http://old.iptel.org/products/cpled/. The page it links to for download
doesn't seem to work anymore.
Does this product still exist?
Thanks for any help.
Doug
--
Doug Meredith
506-854-7997 ext. 801
hi all,
I'd like to include caller and callee SIP contact header in radius
accounting messages. Our dynamic QoS provisioning framework needs this
information to set up QoS per source/destination IP address pair.
Caller contact header is easily done using radius_extra and setting an
AVP in the INVITE handling route block. But I haven't figured out yet
how I could store the callee's contact header in an AVP. The main
problem I have is that I can't use any AVP related functions in
onreply_route blocks. What I'd like to do is something like
modparam("acc", "radius_extra", \
"Originating-Station-IP=$avp(AVP_ORIG_IP); \
Destination-Station-IP=$avp(AVP_DEST_IP)")
onreply_route[ONREPLY_DEFAULT_RELAY] {
if (t_check_status("2[0-9]{2}")) {
avp_write("$ct", "$avp(AVP_DEST_IP)");
}
}
I've learned from an earlier mail thread
(http://www.openser.org/pipermail/users/2005-October/001272.html) that
enabling AVP functions in on_reply routes would be rather difficult. Is
there still a chance that on_reply AVP handling gets implemented?
And is there an alternative method how this could be achieved without
patching the acc module?
thanks,
Christian
Dear OpenSER Users or Admins,
I would like to setup OpenSER as a SIP Redirect Server in my environment.
The point is to redirect depending on the response from DNS or even better
DNS SRV.
The SIP UA will get than as quick as possible the 3xx Response and resend
the request to the redirected address (IP or Domain address) SIP Device.
Is there a possible way to do that via config? How many SIP Request (e.g.
REGISTER, INVITE) will it be possible to redirect per second?
Thanks in Advance for your responses
Oetzi
Hi every one,
I am a new user of the openser.When I am trying to call to a number, the
invite messege is going to proxy server but from proxy, its sending two
invites simultaneously to two phones.
I am attaching my openser.cfg for your kind information.Point to be noted
that, when the phone resisters, no value is being inserted in to the
location table of openser Database.
Thanks in Advance,
Shiva
Please ignore previous post which I sent before completing!
I have two problems:
1. A 2nd Remote-Party-ID header gets appended when I try to do
two subst() on it.
2. Xlog seems to only have a ref to the orignal unedited rpid
header.
I am attempting to clean up my Remote-Party-ID header so that my Polycom
UAs will display the incoming number as a number rather than as a URI.
This requires the domain to match that of the UA's server (e.g.
siptest.columbia.edu). And, I want to rewrite incoming calling numbers
from my PBX to be in 5-digit format and other US 10-digit numbers into
proper E.164 format since my PBX and at least one of my ITSPs doesn't
provide them as E.164.
So, I do a couple of subst() operations on INVITEs. What's wierd is
that one Remote-Party-ID header becomes two when it gets to the UA.
The call flow is:
INVITE r-uri <sip:10508@128.59.59.96:5060> from
<sip:9174147124@128.59.59.242>
With the original RPID header as:
Remote-Party-ID: <sip:9174147124@128.59.59.242>
And after route[10] the headers that end up on the UA are:
Remote-Party-ID: <sip:9174147124@siptest.columbia.edu>
Remote-Party-ID: <sip:+19174147124@128.59.59.242>
It looks like the 2nd subst() happened against the original
RPID, not the edited one and somehow two RPID headers got
created.
Here's the code for the route block:
route[10] {
xlog("L_INFO","route[10] $ci: method $rm r-uri <$ru> from <$fu> rpid
<$re>\n");
# first just clean out the domain name
subst('/^(Remote-Party-ID:.*<sip:.*)@.*>(.*)$/\1(a)siptest.columbia.edu>\2/');
# now strip our numbers down to 5 digits
subst('/^(Remote-Party-ID:.*<sip:)(\+)?(1)?21285([134][0-9]{4})@/\1\2/');
# expand 10-digit NANP numbers to E.164
subst('/^(Remote-Party-ID:.*<sip:)([0-9]{10}.*$)/\1+1\2/');
# ??? xlog doesn't show the result of rewriting it.
# xlog("L_INFO","route[10] RPID is now <$re>\n");
xlog("L_INFO","route[10] $ci: RPID was rewritten.\n");
return;
} #end of route[10]
Any help with this would be greatly appreciated!
/a
dear sir or madame;
i'm trying to start an openser, but i can't, i need to configure properly my
openser.cfg!
if you have an example of how can i configure my openser.cfg, to make it
work, i will very thanksfull
simo
I tried xlog $rs $rr to log the reason for a failure in failure_route
but it just logs NULL. What am I doing wrong? The error status is
available to test with t_check_status and is logged by ACC but not xlog.
# Error from media gateway
failure_route[4] {
xlog("L_INFO","failure_route[4] $si: method $rm r-uri <$ru>\n");
if (t_check_status("(5..)|(6..)|(403)")) { # gateway failure/refusal
xlog("L_INFO","failure_route[4] gateway returned 5xx or 6xx.
Status: $rs $rr\n");
if (!next_gw()) {
xlog("L_CRIT","No more gateways for <$tu>\n");
t_reply("503", "Service not available - No more gateways");
return;
} else {
xlog("L_INFO","failure_route[4] next gw r-uri <$ru> \n");
t_on_failure("4");
t_relay();
return;
}
}
xlog("L_INFO","failure_route[4]: relaying normal error\n");
t_relay();
}
Oct 26 18:02:04 jello /usr/sbin/openser[2028]: ACC: call missed:
method=INVITE, i-uri=sip:9319174147124@siptest.columbia.edu;user=phone,
o-uri=sip:19174147124@67.151.91.22:50
60;transport=udp, call_id=c141e612-75cfa484-cfce783b(a)128.59.37.218,
from="Alan Crosswell"
<sip:10508@siptest.columbia.edu>;tag=CA5BDB76-6DB44EED, code=403 Forbidden
Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4]
128.59.37.218: method INVITE r-uri
<sip:19174147124@67.151.91.22:5060;transport=udp>
Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] gateway
returned 5xx or 6xx. Status: <null> <null>
Oct 26 18:02:04 jello /usr/sbin/openser[2028]: failure_route[4] next gw
r-uri <sip:19174147124@67.151.91.22:5060;transport=udp>
Hi there.
I just getting my Opserser server up and I want to be able to
create child accounts from my system to be able to ring multiple extension
at the same time, is this possible ?
Thanks
Marcos Biscayaqu
Hi,
I want to connect OpenSER with PSTN, as media gateway I use Cisco AS5300, vfc c542.
I am interested about config part from OpenSER.
Where I can get some documentation, information?
Thx
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