Hi list,
my proxy (yy.yy.yy.yy) receives a message like this:
BYE sip:12345678@xx.xx.xx.xx SIP/2.0
Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on>
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0
Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf
Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on>
My ruleset is something like this (with the REGISTER logic pulled out) :
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
route {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
force_rport();
if (!method == "REGISTER") {
record_route();
};
if (loose_route()) {
route(1);
};
loopkup("location");
route(1);
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
exit;
}
%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
I would expect the packet to leave my router, back to xx.xx.xx.xx :
BYE sip:12345678@xx.xx.xx.xx SIP/2.0
Record-Route: <sip:yy.yy.yy.yy;lr;ftag=2873864>
Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on>
Via: SIP/2.0/UDP yy.yy.yy.yy;branch=zu83fdhDB8d.67235676.0
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bK07cb.466d8587.0
Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf
Route: <sip:xx.xx.xx.xx;ftag=1056999653;lr=on>
But the outgoing packet is :
BYE sip:xx.xx.xx.xx;ftag=1056999653;lr=on SIP/2.0
Record-Route: <sip:xx.xx.xx.xx;ftag=875188d2;lr=on>
Via: SIP/2.0/UDP yy.yy.yy.yy;branch=z9hG4bK07cb.02005244.0
Via: SIP/2.0/UDP xx.xx.xx.xx;rport=5060;branch=z9hG4bK07cb.466d8587.0
Via: SIP/2.0/UDP ww.ww.ww.ww:5060;branch=z9hG4bK875196bf
Do you have any idea about what is going on ?
Thank you.
Best regards,
--
Jeremie Le Hen
< jeremie at le-hen dot org >< ttz at chchile dot org >
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hi list,
we have some trouble handling NAT correctly when receiving a call from
another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP
when it finds another proxy's record-route header, though it invokes
mediaproxy and sends uncorrected SDP to the other SIP-server resulting
in one-way-audio as the provided c=-field includes the RFC1918-address
of the client.
when receiving calls through our Cisco AS5350 everything is fine
('cause it does not add the record-route). the addition of the header
cannot be circumvented bacause the other provider wants to bill calls
(obviously) and therefor needs to be in the SIP-dialog.
is there a neat solution to this problem or do we have to get a
session-border-controller or some B2BUA-config for handling the
received calls?
and by the way, how do i distinguish external INVITEs destined for one
of our clients (uri==myself) from the INVITEs being sent out to the
callee upon an external INVITE? if there was a way we could invoke
mediaproxy from there? how is the recommended setup?
brgds,
g.feiner
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Hello.
openser 1.1
radiusclient-ng 5
freeradius 1.1.1
I need to enable the modules *_radius for my openser
instalation a follow the guide for enable radius then
when i try to start openser i get the following error.
0(0) ERROR: auth_radius: can't get code for the
Digest-Response attribute
Anyone knows what is the cause for this error?
Thank's
Alex.
___________________________________________________________
Do You Yahoo!?
La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx
Hi,
Is there a way to coexist multiple domains in Ser, distinguishing users
even if they have the same username in subscriber table but different
domains?
I'll explain better, I have subscriber table with:
| username | domain | e-mail |
------------------------------------------------
| user_A | domain_A | user_A@domain_A |
------------------------------------------------
| user_A | domain_B | user_A@domain_B |
and in domain table there are both domains domain_A and domain_B.
But when someone calls user_A, (with or without writing their domain), both phones ring (if both users have their phones
registered), when I want only one to ring because they are different users!!
Is there a way in Ser to do this right?
Regards,
Ricardo.
It's true Asterisk can use different contexts, but that doesn't solve my
problem because I use Ser + Asterisk and Ser is used for routing SIP
messages between SIP UAs, not Asterisk. In my architecture, Asterisk is
only used for advanced services like voicemail or call conferencing and
for gateway, not to route messages between UAs.
Ricardo.
openser(a)comcast.net wrote:
> The way to do that is intergrate Ser with Asterisk and in Asterisk you
> can build context and keep the same numbers for different context like
> a pbx.
>
>
> -------------- Original message --------------
> From: Ricardo Carvalho <rcarvalho(a)iric.up.pt>
>
> > Hi,
> >
> > Is there a way to coexist multiple domains in Ser,
> distinguishing users
> > even if they have the same username in subscriber table but
> different
> > domains?
> >
> > I'll explain better, I have subscriber table with:
> >
> > | username | domain | e-mail |
> > ------------------------------------------------
> > | user_A | domain_A | user_A@domain_A |
> > ------------------------------------------------
> > | user_A | domain_B | user_A@domain_B |
> >
> > and in domain table there are both domains domain_A and domain_B.
> >
> > But when someone calls user_A, (with or without writing their
> domain), both
> > phones ring (if both users have their phones
> > registered), when I w! ant onl y one to ring because they are
> different users!!
> > Is there a way in Ser to do this right?
> >
> > Regards,
> >
> > Ricardo.
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > Serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
Hi
I have existing SER Running
i have local users ( extensions) and DID ( all other countries)
iam able to make calls in and out , there is no problem
but when the user not available, it has to send to Voice mail, but its only
sending the users
who are locally start with 9*,
how can i send the user start with 9 or any DID not availble send to
voicemail
here is my config
if (!lookup("location") &&
( uri=~"^sip:9.*@domain.com" )
) {
# Voicemail specific configuration - begin
if(method=="ACK" || method=="INVITE" ||
method=="BYE"){
setflag(2);
if(t_newtran()){
t_reply("100","Trying -- just wait a minute
!");
if(method=="INVITE"){
log(1,"**************** vm start -
begin ******************\n");
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the answer
machine\n");
t_reply("500","could not contact the
answer machine");
};
log(1,"**************** vm start - end
******************\n");
break;
};
if(method=="BYE"){
log(1,"**************** vm end - begin
******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer
machine\n");
t_reply("500","could not contact the
answer machine");
};
log(1,"**************** vm end - end
******************\n");
break;
};
}
else {
log("could not create new transaction\n");
sl_send_reply("500","could not create new
transaction");
};
break;
};
break;
};
any help will be great
Ram
Can somebody please take a look at the routing logic below...
I am trying to use exec_dset along with another program to do some LCR
with SER but it doesn't work properly (here is a capture of what it
produces: http://pastebin.ca/146771
Any ideas of what's doing wrong? Code:
# --------------------------------------
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
break;
};
if (method=="REGISTER") {
sl_send_reply("501", "Not Implemented");
break;
};
if (method=="OPTIONS") {
sl_send_reply("501", "Not Implemented");
break;
};
record_route();
setflag(1);
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
if (method=="INVITE") {
route(2);
break();
};
route(1);
break();
}
route[1]
{
if (!t_relay()) {
sl_reply_error();
};
}
route[2]
{
# replaces the INVITE URI with the right host
exec_dset ("/usr/local/Asterisk-LCR/bin/ser-lcr");
if (uri=~"^sip:STOP@127") {
sl_send_reply("503", "Service Unavailable");
break;
};
t_on_failure("2");
if(isflagset(1)) {
append_branch();
};
setflag(1);
t_relay();
}
# /usr/local/Asterisk-LCR/bin/ser-lcr will give us the next uri to try
# in case of failure, or send sip:STOP@127.0.0.1 if no more routes
failure_route[2] {
route(2);
}
# --------------------------------------
Hi,
I am new to SER and need experts' help in PSTN call routing block to call
termination service provider in a scenario below:
We have our own sip realm/server that will handle all VoIP calls among our
own subscribers; sip.ours.com. This will forward all PSTN calls,
authenticated and authorized by it, to sip.theirs.com, which will
authorize usning our IP address and forward to pstngateway.theirs.com.
What I have been given by SP is:
sip.theirs.com, IP, myuserID and password.
What they are given by me is our IP and hostaname.
sip.ours.com and IP.
My first try was simply;
rewritehostport(sip.theirs.com:5060);
t_relay();
,which fails.
Can someone tell me proper approach in ser commands based on the scenario
above?
Thank you so much in advance.
John K
Hi Bill,
Bill Zhang wrote:
>Anyone can explain following options:
>
>1. -2fv
>
>
v - print the version
f - stay in foreground, don't become a daemon
2 - duplicate mode
>2. -s path
>
>
command socket path
>3. -L nfiles
>
>
maximum file descriptor number that can be opened
regards,
bogdan
>The rest of them seems to be straight forward. This rtproxy documentation
>is HARD to find:(.
>
>usage: rtpproxy [-2fv] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] [-t
>tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles]
>
>Thanks in advance.
>
>Best Regards,
>Bill
>
>
