Hi list,
My SER sends to a specific GW all the external calls, but I intend to have a secondary (backup) GW to terminate those calls would't be completed for some reason.
Does nbody knows how?
Thanks,
Vitor Brasileiro.
Hello List
I have a question regarding the client_nat_test() in the mediaproxy
module.
If i have an INVITE something like this.
INVITE sip:5555848114@mydomain.com SIP/2.0.
Via: SIP/2.0/UDP sipgw.mydomain.com:5060;
branch=z9hG4bKNxvDQVViBAAJAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA_.
Via: SIP/2.0/TCP xx.xx.154.68:14123.
From: <sip:22408100@sipgw.mydomain.com>;tag=3655298548.
To: sip:5555848114@mydomain.com.
CSeq: 1 INVITE.
Call-ID: 3406152498(a)sipgw.mydomain.com.
Contact: <sip:22408100@sipgw.mydomain.com>.
Accept: application/sdp.
Max-Forwards: 69.
Content-Type: application/sdp.
Content-Length: 200.
.
v=0.
o=sipgw.mydomain.com 811562435 1354711274 IN IP4 xx.xx.154.36.
s=SIP Library call.
c=IN IP4 xx.xx.154.68.
t=3312294794 0.
m=audio 23264 RTP/AVP 18 4.
a=rtpmap:18 G729/8000.
a=rtpmap:4 g723/8000.
How is possible to mediaproxy module detect this client as a NAT'd? Because
all the calls coming from this GW are being proxied by mediaproxy.
It suppose that this GW's has a public IP, so i don't understand why is this
happening.
Beside this can someone tell me if this is ok that a VIA parameter has a
domain name ????
The snippet for my NAT'd mediaproxy enviroment is :
if (client_nat_test("3")) {
log(1, "NAT: Requerimiento de IP privada --> fixed contact
(en rutina principal)\n");
setflag(5);
force_rport();
fix_contact();
append_hf("P-hint: fixed NAT contact for request\r\n");
....
if (method=="INVITE") {
log(1, "ROUTING: Reply processing (online user) enabled to
handle NAT'd resonses\n");
t_on_reply("1");
if ( (isflagset(5)) && (method=="INVITE" ||
method=="ACK") ) {
log(1, "NAT: Invite received --> enabling
media proxy (else del lookup 2do if)\n");
use_media_proxy();
append_hf("P-hint: request forced to media
proxy\r\n");
};
};
....
onreply_route[1] {
# If we've got here, it's because we've previously passed through a
block
# which handles NAT requests and has set a t_on_reply condition. DB
03-08-2004
if (status =~ "(183)|(2[0-9][0-9])") {
if (client_nat_test("3") || isflagset(5)) {
log(1, "NAT: Reply from NAT'd client --> fixing contact
(onreply_route)\n");
fix_contact();
use_media_proxy();
setflag(5);
};
log(1, "NAT: NAT'd transaction answered --> enabling media proxy
(onreply_route)\n");
# use_media_proxy();
};
}
Thanks in advance.
Best reegards,
Ricardo.-
Hello!
Finally I got check_to() working well with the uri table. However I
discovered, that there is a problem with some SIP UAs which does not
support or not work well with SRV records.
What I mean:
I have a domain: example.com
My SIP UA doesn't support SRV records, so I have to configure it
directly to use my SIP server. Therefore the UA sends uri like
xyz(a)hostname.example.com.
My SER is configured to use
if (method=="REGISTER") {
if (!www_authorize("example.com", "subscriber")) {
www_challenge("example.com", "0");
break;
};
if (!check_to()) {
sl_send_reply("403", "Registration id must does not fit");
break;
};
save("location");
break;
};
In logs I see:
0(15538) HA1 string calculated: 589858581b9363bcd477f922ff71172a
0(15538) check_response(): Our result = 'b3fcd7f8f83981c994ff52f388001415'
0(15538) check_response(): Authorization is OK
0(15538) save_rpid(): rpid value is ''
0(15538) check_username(): Digest realm and URI domain do not match
0(15538) parse_headers: flags=-1
What is the most elegant way to solve such case - together with UAs
which can use SRV? Any idea, hints?
Thanks in advance,
Tamas
Thanks for the script, I did have one question. Your script mentions using a column labeled "timestamp" using UNIX_TIMESTAMP. I'm assuming of course this is, well, unix time :-) I couldn't find anywhere in the acc module docs to output "CDR's" in unixtime, did you have something attached to SER to convert it? Thanks!
Matt
-----Original Message-----
From: Darren Nay [mailto:dnay@IONOSPHERE.net]
Sent: Thursday, December 16, 2004 11:59 AM
To: serusers(a)lists.iptel.org
Subject: RE: [Serusers] Does SER do that?
Sure, I am happy to share it.
Keep in mind. This script assumes that you are using a phone number (DID) username scheme. So if your usernames aren't in a standard DID format of 1XXXXXXXXXX or +1XXXXXXXXXX then you might have to tweak it some to make it work correctly for you.
Give it a try and give me some feedback. Thanks!
Darren Nay
> -----Original Message-----
> From: Matt Schulte [mailto:mschulte@netlogic.net]
> Sent: Thursday, December 16, 2004 11:58 AM
> To: Darren Nay; Alberto Martínez; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Does SER do that?
>
> Nice! Care to share the app? :-) heehee
>
> -----Original Message-----
> From: Darren Nay [mailto:dnay@IONOSPHERE.net]
> Sent: Thursday, December 16, 2004 10:39 AM
> To: Matt Schulte; Alberto Martínez; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Does SER do that?
>
>
> Yes, this can be done to some degree.
>
> I've written a perl script that queries the "acc" table and keeps
> track of current calls. The only issue is that occasionally if a call
> is not broken down correctly (ie. SER did not receive the BYE) then
> you may have some calls reported to stay up when actually they are
> not.
>
> However, the best way to keep track of concurrent calls is to use
> mediaproxy. It always knows exactly how many calls are up because it
> carries the audio for each call. Much more accurate than using sers
> "acc" table.
>
> That being said, we still use the "acc" table method because we prefer
> that the audio streams not ride our network. It saves cost on
> bandwidth and we've found that the voice quality is better directly
> from IAD to PSTN/IAD and not relayed through an RTP proxy.
>
> Darren
>
>
> > -----Original Message-----
> > From: Matt Schulte [mailto:mschulte@netlogic.net]
> > Sent: Thursday, December 16, 2004 11:38 AM
> > To: Alberto Martínez; serusers(a)lists.iptel.org
> > Subject: RE: [Serusers] Does SER do that?
> >
> > Yes, yes, and maybe.
> >
> > > I would like to know if SER can notify by anyway to other
> > > application
> > of the start and end of
> > > VoIP transmissions in order to maintain a control in real-time of
> > > calls
> > made by the users.
> >
> > This one maybe a little more tricky, I'm looking for such an app
> > myself. I suppose this could be done via the acc module.
> >
> > -----Original Message-----
> > From: Alberto Martínez [mailto:amartinez@astrasoft.es]
> > Sent: Thursday, December 16, 2004 9:21 AM
> > To: serusers(a)lists.iptel.org
> > Subject: [Serusers] Does SER do that?
> >
> >
> > -----BEGIN PGP SIGNED MESSAGE-----
> > Hash: MD5
> >
> > Hello,
> >
> > I am new in SER. I would like to ask you if SER is able to
> > do somethings I am looking for. I would like it to check if the
> > users who try to connect with it by SIP are correct, checking the
> > login info and, if they are, redirect the connection to the VoIP
> > provider. If the user who is trying to connect is not set up
> > in SER it must be rejected.
> >
> > I would like to know if SER can notify by anyway to other
> > application of the start and end of VoIP transmissions in order
> > to maintain a control in real-time of calls made by the users.
> >
> > Thank you.
> >
> > Best regards,
> > Alberto
> >
> > -----BEGIN PGP SIGNATURE-----
> > Version: 2.6
> >
> > iQEVAwUAQcGn+HHoJ4bX5QlXAQEyCAf/T1bEaVlXWW+krVKlFl5yxJLLv3uFH9q0
> > qddg/6+YDx1lXnoigqxTWtbpvR0uMnVnFRqueifpWltzkawcgHYtVRNmHV1+bpDn
> > awrgE0UFb2zlaUsFf+INqaaFuXGrztgCA0jcwh4gvhAYzQX8L/32g9EsRrZNNpie
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> > =tOcm
> > -----END PGP SIGNATURE-----
> >
> >
> > _______________________________________________
> > 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
Yes, this can be done to some degree.
I've written a perl script that queries the "acc" table and keeps track of
current calls. The only issue is that occasionally if a call is not broken
down correctly (ie. SER did not receive the BYE) then you may have some
calls reported to stay up when actually they are not.
However, the best way to keep track of concurrent calls is to use
mediaproxy. It always knows exactly how many calls are up because it
carries the audio for each call. Much more accurate than using sers "acc"
table.
That being said, we still use the "acc" table method because we prefer that
the audio streams not ride our network. It saves cost on bandwidth and
we've found that the voice quality is better directly from IAD to PSTN/IAD
and not relayed through an RTP proxy.
Darren
> -----Original Message-----
> From: Matt Schulte [mailto:mschulte@netlogic.net]
> Sent: Thursday, December 16, 2004 11:38 AM
> To: Alberto Martínez; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Does SER do that?
>
> Yes, yes, and maybe.
>
> > I would like to know if SER can notify by anyway to other application
> of the start and end of
> > VoIP transmissions in order to maintain a control in real-time of calls
> made by the users.
>
> This one maybe a little more tricky, I'm looking for such an app myself. I
> suppose this could be done via the acc module.
>
> -----Original Message-----
> From: Alberto Martínez [mailto:amartinez@astrasoft.es]
> Sent: Thursday, December 16, 2004 9:21 AM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Does SER do that?
>
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: MD5
>
> Hello,
>
> I am new in SER. I would like to ask you if SER is able to do
> somethings I am looking for. I would like it to check if the users who try
> to connect with it by SIP are correct, checking the login info and, if
> they are, redirect the connection to the VoIP provider. If the user who
> is trying to connect is not set up in SER it must be rejected.
>
> I would like to know if SER can notify by anyway to other application of
> the start and end of VoIP transmissions in order to maintain a control
> in real-time of calls made by the users.
>
> Thank you.
>
> Best regards,
> Alberto
>
> -----BEGIN PGP SIGNATURE-----
> Version: 2.6
>
> iQEVAwUAQcGn+HHoJ4bX5QlXAQEyCAf/T1bEaVlXWW+krVKlFl5yxJLLv3uFH9q0
> qddg/6+YDx1lXnoigqxTWtbpvR0uMnVnFRqueifpWltzkawcgHYtVRNmHV1+bpDn
> awrgE0UFb2zlaUsFf+INqaaFuXGrztgCA0jcwh4gvhAYzQX8L/32g9EsRrZNNpie
> WK8uYxp7N9Pw6MqkQwvSrCrbuIt+umP+tbYJcza83d5+Bb/yNXn8ePY1ztnWYfZ+
> 5vp8jimNO/93T9/k26zs1hdGEtW68tCIbMeWu37FFmPbRthlGQM/a5Ku76ZfhjjO
> Lmv9fi3spkH+uOfgjlX6YFLdgEW1bxk7bEL0yPf7hB+2Awgqy3QIfw==
> =tOcm
> -----END PGP SIGNATURE-----
>
>
> _______________________________________________
> 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
Hello,
I have a problem authentication my ATA devices with mysql , I have this configuration; The configuration in my sjphone is working.
SJPHONE: 192.168.10.90
ATA IP: 192.168.10.96
SER: 192.168.10.91
SIP_DOMAIN="192.168.10.91"
ser.cfg
alias="192.168.10.91"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
modparam("usrloc", "db_mode", 2)
if (!www_authorize("192.168.10.91", "subscriber")) {
www_challenge("192.168.10.91", "0");
break;
};
ATA version 3.2
UID0: 911211389
PWD0: bolo
UseLoginID: 1
Proxy: 192.168.10.91
SJPHONE
Proxy: 192.168.10.91
Domain: 192.168.10.91
Account: 911211390
Password: bolo
MYSQL
serctl add 911211389 bolo 911211390(a)192.168.10.91
serctl add 911211390 bolo 911211390(a)192.168.10.91
And this is why I have in my ethereal
SJPHONE --> SER
sip.Request-Line = Request-Line: REGISTER sip:192.168.10.91 SIP/2.0x
sip.From = From: <sip:911211390@192.168.10.91>;tag=147639828
sip.To = To: <sip:911211390@192.168.10.91>
sip.Authorization = Authorization: Digest username="911211390",realm="192.168.10.91",nonce="41c046cac4f98869173e6aa7a2bb4517a7f7e9d6",uri="sip:192.168.10.91",response="2751aa78ee74298024815ba8cf35f2a7"
SER --> SJPHONE
sip.Status-Line = Status-Line: SIP/2.0 200 OK
sip.From = From: <sip:911211390@192.168.10.91>;tag=147639828
sip.To = To: <sip:911211390@192.168.10.91>;tag=8e7dfe287ff1fc069b31517361891514.a624
sip.Warning = Warning: 392 192.168.10.91:5060 "Noisy feedback tells: pid=2409 req_src_ip=192.168.10.90 req_src_port=5060 in_uri=sip:192.168.10.91 out_uri=sip:192.168.10.91 via_cnt==1"
ATA --> SER
sip.Request-Line = Request-Line: REGISTER sip:192.168.10.91 SIP/2.0
sip.From = From: 911211389 <sip:911211389@192.168.10.91;user=phone>;tag=2633904518
sip.To = To: 911211389 <sip:911211389@192.168.10.91;user=phone>
sip.Authorization = Authorization: Digest username="911211389",realm="192.168.10.91",nonce="41c054a27a512a571fd108a7cb2c15aaba345768",uri="sip:192.168.10.91",response="924c72dcd60682cbe463a1dbe5521581"
SER --> ATA
sip.Status-Line = Status-Line: SIP/2.0 401 Unauthorized
sip.From = From: 911211389 <sip:911211389@192.168.10.91;user=phone>;tag=2633904518
sip.To = To: 911211389 <sip:911211389@192.168.10.91;user=phone>;tag=8e7dfe287ff1fc069b31517361891514.2cd2
sip.WWW-Authenticate = WWW-Authenticate: Digest realm="192.168.10.91", nonce="41c054a27a512a571fd108a7cb2c15aaba345768"
sip.Warning = Warning: 392 192.168.10.91:5060 "Noisy feedback tells: pid=2391 req_src_ip=192.168.10.96 req_src_port=5060 in_uri=sip:192.168.10.91 out_uri=sip:192.168.10.91 via_cnt==1"
I donno why my sjphone is registring withoput problems and my ATA is rejected, any help will be nice
Thx in advance.
I've just created the rel_0_9_0 ser cvs branch.
This release is a testing release, is not meant to replace stable (yet).
So don't switch all your productions systems to it :-)
When we'll get confident enough we'll have a new stable release from
this new branch.
You can either download a source tar.gz from
ftp://ftp.berlios.de/pub/ser/0.9.0/src/
or use cvs:
cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co -rrel_0_9_0 sip_router
cvs is recommended since you'll get all the bug fixes (the tar.gz will
not be updated so often).
Unfortunately the documentation has not been updated yet.
See NEWS for a very brief description of what changed
(http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/NEWS?rev=1.38&only… ).
Andrei
i am a student n wishing to install ser on my PC it is having fedora core2.
kindly guide me how to do it,
where to get the rpm's from,
do i require to change the OS to red hat 9 or something else,
how to download the complete package from ftp and so on....
sush
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