Anybody noticed that the site iptel.org/ser has a new look and the ser
admin and ser programmer guides (among many other links) are gone.
Are these doc still accessible ??
Thanks
Gaurav
Hi,
Oh my. My appologies! I've mixed up with Asterisk.
What ser version do you want to install, btw? If 0.9.x then plz take a
look at INSTALL file in the tar ball. There should not be any problem
with compiling like in any other linux system, if you've installed all
the needed packages.
When you make,
1. make #builds only ser core, equivalent to make ser
2. make all #builds everything
3. make bin include_modules="mysql jabber cpl-c auth_radius
group_radius uri_radius postgres pa" #builds …
[View More]selective modules
(you'll definitely need mysql and the rest is up to you)
4. Refer to INSTALL file for more options. Btw, since Ubuntu is Debian
based, then also take a look at the "requirements" section for Debian.
5. Make sureto install MySQL first. I used 4.1.x for many
installations so far and it seems to be working really fine.
Then make install. This should install ser into default directory (
/usr/local ) with configuration file inside the /usr/local/etc/ser/
and shell scripts with binaries in /usr/local/sbin/.
Andrey.
On 8/19/06, Eddie Johnson Jr <ilo(a)ilo-infosystems.us> wrote:
> Are you saying in this e-mail that a package exist in the repository because
> I have checked by searching SER and Sip Express Router and nothing. Any
> suggestions will be greatly appreciated.
>
> -----Original Message-----
> From: serusers-bounces(a)lists.iptel.org
> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov
> Sent: Thursday, August 17, 2006 10:57 PM
> To: serusers(a)iptel.org
> Subject: Re: [Serusers] SER on Ubuntu Breezy
>
> Hi,
>
> A friend of mine has installed SER on Ubuntu 5.10 before from Ubuntu's
> package list. I dont know the details, but it should be fairly easy.
> Flex and bison MUST be installed on Ubuntu by default. All u need to
> install is MySQL server first, and then SER. Ubuntu should take care
> of the rest or at least tell u which package is missing.
>
> On 8/17/06, Eddie Johnson Jr <ilo(a)ilo-infosystems.us> wrote:
> >
> >
> >
> > Has anyone done an install of SER on Ubuntu 5.10. I need some initial
> setup
> > points and I can take it from there. I installed, flex, bison,
> > build-essential, ran make clean nothing, ran make install and I get
> makefile
> > 80 no directory or file. Where do I go from here?
> >
> >
> >
> > Regards,
> >
> >
> >
> > iloadmin
> > _______________________________________________
> > Serusers mailing list
> > Serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> >
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
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Hey all..
First time poster, but long time reader. Thanks to all that contribute
to this excellent project, this software is very flexible!
I just wanted to mention that while upgrading my config to 1.1.0, I
found that..
avp_write("$ruri/username", "$avp(myuser)") or
avp_write("$ru/username", "$avp(myuser)")
does not work as expected, and does not return just the username.
Instead I needed to change it to..
avp_write("$rU", "$avp(myuser)")
BTW, am I able to update via SVN …
[View More]to a newer bug fixed version of the
1.1.0 branch, without taking on any new things in the HEAD branch? Is
it just as simple as this, and I have just not tried it. I wish to use
this code in a production environment.
Kind regards, Stuart.
[View Less]
Hello Greger and thanks for your answer
I tested sending the CANCEL to route(3), and it seems to be working ok now. But now i'm facing another problem. The OK message for the BYE seems not to be routed back to the NAT'd port, instead is routed to the default port. Please check the file attached (BYE_debug.TXT).
As i pointed in my first post, the calling endpoint is NAT'd. Following the BYE sequence seems to be right, it hits the loose_route section and then the route(1), but since …
[View More]the BYE does not contain the rport parameter in the VIA header the OK is routed back to the port advertised in the VIA header. (5061) and not to the received port.
Maybe i'm doing something wrong?.
I tested a new solution, but i don't know if is correct or is according to the ONSIP document.
I added a client_nat_test("3") in the Call Tear Down Section-
..........
# -----------------------------------------------------------------
# Record Route Section
# -----------------------------------------------------------------
if (method=="INVITE" && client_nat_test("3")) {
# INSERT YOUR IP ADDRESS HERE
record_route_preset("200.100.100.246:5060;nat=yes");
} else if (method!="REGISTER") {
record_route();
};
# -----------------------------------------------------------------
# Call Tear Down Section
# -----------------------------------------------------------------
if (method=="BYE" || method=="CANCEL") {
if (client_nat_test("3")) {
force_rport();
xlog("L_INFO", "[%rm] - From:%fu ; To:%ru ; Call-ID:%ci ; Desde:%is\n - FORCE RPORT ON");
};
end_media_session();
setflag(1);
};
# -----------------------------------------------------------------
# Loose Route Section
# -----------------------------------------------------------------
if (loose_route()) {
if (has_totag() && (method=="INVITE" || method=="ACK")) {
if (client_nat_test("3")||search("^Route:.*;nat=yes")){
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
.......
Hope that someone could help me here.
Thanks
Regards,
Ricardo Martinez.-
-----Mensaje original-----
De: Greger V. Teigre [mailto:greger@teigre.com]
Enviado el: jueves, 17 de agosto de 2006 4:05
Para: Ricardo Martinez
CC: serusers(a)iptel.org
Asunto: Re: [Serusers] Via RPORT for CANCEL message.
Having a quick look, you may have found a bug in the ONsip.org scripts. A CANCEL is not loose routed (a dialog has not yet been set up) and must be handled the same way as an INVITE, i.e. the only way to find the route for CANCEL is to route the CANCEL the same way you routed the INVITE. However, it seems that you have somehow has done that, as your domain has changed to @gw. in the CANCEL doing downstream (i.e. forwarded by your SER).
So, CANCEL should be sent to route(3), the INVITE handler and not route(1). I'm quite sure this was correct in a previous version, we even had some discussions around how to handle it and how to describe it, but it seems that something has happened along the way.
g-)
Ricardo Martinez wrote:
Hello.
My main configuration is based on the ser.cfg from the issue 5 plus some modifications included by me. Beside that i'm using
version: ser 0.9.3 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $
main.c compiled on 11:03:37 Dec 22 2005 with gcc 3.2
I have a question regarding to sending a CANCEL message for an INVITE from a NAT'd endpoint. As far as i know the force_rport() command adds the received IP port to the top most via header in the SIP message, this enables subsequent SIP message to return to the proper port later on in a SIP transaction. My problem is that for a CANCEL message coming from a NAT'd endpoint this command seems not to be working. This is the scenario.
NAT'd endpoint : 200.100.100.248
SER : 200.100.100.246
SER-2 : 200.100.100.36
GW : 200.100.100.69
The NAT'd endpoint send an INVITE to the proxy, but then in the middle of the transaction decide to CANCEL the request.
As you can see in the "cancel_debug.txt" file included on this mail, the CANCEL message does not contain the "rport" in the Via header, so it seems to be routed back to the default sip port (5060). Is there a way to force the rport in a CANCEL? for a enpoint unable to put the rport by itself?
I want to do this according to the onsip document, in a compatible way..
Anyway i made a little test including a force_rport(); in the CANCEL handler
..........
# -----------------------------------------------------------------
# Record Route Section
# -----------------------------------------------------------------
if (method=="INVITE" && client_nat_test("3")) {
# INSERT YOUR IP ADDRESS HERE
record_route_preset("200.100.100.246:5060;nat=yes");
} else if (method!="REGISTER") {
record_route();
};
# -----------------------------------------------------------------
# Call Tear Down Section
# -----------------------------------------------------------------
if (method=="BYE" || method=="CANCEL") {
end_media_session();
setflag(1);
};
# -----------------------------------------------------------------
# Loose Route Section
# -----------------------------------------------------------------
if (loose_route()) {
if (has_totag() && (method=="INVITE" || method=="ACK")) {
if (client_nat_test("3")||search("^Route:.*;nat=yes")){
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
# -----------------------------------------------------------------
# Call Type Processing Section
# -----------------------------------------------------------------
if (!is_uri_host_local()) {
if (is_from_local() || allow_trusted()) {
route(4);
route(1);
} else {
sl_send_reply("403", "Forbidden");
};
break;
};
if (method=="CANCEL") {
force_rport();
route(1);
break;
} else if (method=="INVITE") {
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
} else if (method=="ACK") {
route(1);
break;
};
lookup("aliases");
if (uri!=myself) {
route(4);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(1);
this solve my problem but it seems not to be so accurate.
I hope that someone can help me
Thanks!
Ricardo Martinez.-
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Hi everybody!
I am running SER + RTPPROXY and I am trying to establish a call between a UA
with public IP address and a UA with private IP address connecting to the proxy
through a Access Point that has an extern ip address that is private too:
UA1 (192.168.0.100) <--> (192.168.0.1) AP (192.168.1.1) <--> (192.168.1.2) SER
(public IP) <--> (public IP) UA2
I just used the function force_rtp_proxy of nathelper module to force SER to
change the sdp body of INVITE and …
[View More]reply messages, and in this way the fields c=
and m= of the sdp get changed correctly.
I can establish a call and the UAs send RTP packets to the rtpproxy. The
problem is that the rtpproxy doesn't forward the RTP packets coming from the UA
with public address (UA2) towards the UA with private address (UA1).
Are there special flags I have to set in order to make rtpproxy work properly?
Do I have to use other functions of nathelper module besides force_rtp_proxy? I
am writing my own module for SER and I programmed it in such a way that the
force_rtp_proxy function is called (not from the configuration file but from
the module itself) for every INVITE request and reply, so that every RTP stream
has to pass through the rtpproxy.
If I try to use 2 UAs with public addresses everything works properly...
Last question: how can I run rtpproxy in debug mode? And where can I see the
log files???
Please someone help me!
Thanks a lot!
Chiara
[View Less]
I have just upgraded from 1.0.0 to 1.1.0-snapshot of a couple days ago,
and now I get the following errors:
Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcp_blocking_connect:
timeout 10 s elapsed from 10 s
Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcpconn_connect:
tcp_blocking_connect failed
Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR: tcp_send: connect failed
Aug 18 14:50:56 eowyn OpenSER[16760]: msg_send: ERROR: tcp_send failed
Aug 18 14:50:56 eowyn OpenSER[16760]: ERROR:tm:…
[View More]t_forward_nonack: sending
request failed
I'm using OpenBSD 3.9-stable amd64.
What could be the problem?
Thanks.
P.S.
It could be a good idea to add in the 1.0.x to 1.1.x upgrade web page
that the "module_exports" structure has changed.
P.S. 2
My system is a 64 bit one, but the standard library directory is the
usual "lib" and NOT "lib64".
--
___________________________________________________
__
|- giannici(a)neomedia.it
|ederico Giannici http://www.neomedia.it
___________________________________________________
[View Less]
Sure
In sip.conf create a [user] and a [peer] or you can use [friend] if the
para's are the same
[SER]
type=user
disallow=all
allow=g729
allow=ulaw
allow=alaw
context=default
canreinvite=no
permit=10.0.0.2/255.255.255.255
host=10.0.0.2
trustrpid=no
[SER]
type=peer
insecure=yes
host=10.0.0.2
context=default
nat=no
canreinvite=no
promiscredir=yes
disallow=all
allow=g729
allow=ulaw
trustrpid=no
Rick
_____
From: serusers-bounces(a)lists.iptel.org
[…
[View More]mailto:serusers-bounces@lists.iptel.org] On Behalf Of Daniel Mirrha Santos
Sent: Friday, August 18, 2006 8:48 AM
To: Ser
Subject: [Serusers] Connect Ser on a Asterisk
Hi,
I'm using SER like a public switch and I need to register it on a Asterisk
server. Does anybody know how could I do that?
Regards,
Daniel
[View Less]
I think Ser also can fire some kind of interruption that calls some
routine when the call rings for some time. This routine being called
should send that call to voicemail.
Anyone knows how can I do that in ser.cfg?
Thanks,
Ricardo.
(Also thanks Juan for your post!)
Juan wrote:
> Hi Ricardo.
> SER doesn't care if the call is answered or not.
> Asterisk yes. From the little I know about * you will have to setup a
> mailbox for each customer so if he/she is not online or …
[View More]doesn't answer the
> call it will be redirected after certain amount of time to the mailbox.
> Hope this helps.
> Regards
>
> Juan
>
> -----Original Message-----
> From: serusers-bounces(a)lists.iptel.org
> [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Ricardo Carvalho
> Sent: Thursday, August 17, 2006 1:33 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] How to send Calls from Ser to Asterisk Voicemail?
>
> How should I configure ser.cfg to enable Ser to send calls to Asterisk
> Voicemail when after ringing for some while, the phone isn't answered?
>
> Thanks,
>
> Ricardo.
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
[View Less]
Hi,
I would like to know is it possible to setup SER with multiple media proxy. If its not possible than please suggest me what is the best way to solve the problem mentioned below.
The problem is I have a setup of SER with Media proxy running in Singapore and I also have some users in US. But I want to use one SER, which is running in singapore, since it connects with DB and doing the AAA. And my users in US also register on the same proxy. But when my US users make IP to IP calls since …
[View More]in current setup their media is first coming to singapore than going to US, they face a latency and poor quality. How can I avoid this without running multiple instances of SER.
What I plan to do is setup another media proxy in US and depending on the Nated contact of user agents I give them appropriate media proxy address.
Thanking you in advance,
Best Regards,
Abdul Qadir
Best Regards,
Abdul Qadir
---------------------------------
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