Hi!
I wonder how the NAT flag will be set during lookup("location") in case
of multiple branches and some of the contacts are behind NAT and some not.
regards,
klaus
Hi all,
I'm still at dbtext:
I have a line in ser.cfg which is now:
modparam("auth_db", "db_url","dbtext:///usr/local/etc/ser/dbtext")
where I have a file produced with the script from cesc called "subscriber".
Given by DEBUG-mode from dbt_load_file the module "dbtext" searches this file in a
directory called "/usr/local/etc/ser" and NOT in the dir given by the "modparam"-line
Can somebody give me a hint, why this occurs?
Another problem I had with the strncpy/strncat of database and table-name.
I had to change a line in db_file.c at line 82 from
strncpy(path+dbn->len+1,tbn->s,tbn->len)
to
strncat(path,tbn->s,tbn->len)
Is this a bug or is something wrong with my ser.cfg?
Thanks Thorsten
ser-version: 0.8.14
mt> -----Ursprüngliche Nachricht-----
mt> Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
mt> Gesendet: Freitag, 29. Juli 2005 10:55
mt> An: Müller Thorsten
mt> Cc: serusers(a)lists.iptel.org
mt> Betreff: Re: [Serusers] dbtext: table not loaded. example desired
mt>
mt>
mt> Hello,
mt>
mt> On 07/29/05 11:16, Müller Thorsten wrote:
mt>
mt> > Hi all,
mt> >
mt> > I try to set up ser with dbtext as authentication basis.
mt> I run in some
mt> > problems because the database is not found:
mt> > DBT:db_query: table not loaded.
mt> >
mt> > After some googling a found that Cesc
mt> [cesc.santa(a)gmail.com] had also
mt> > done some work with dbtext and mailed that to "serusers"
mt> or "serdev"
mt> > mailinglists
mt> >
mt> > In a mail from May and June 05 he told that he has
mt> written a "serctl"
mt> > for dbtext and a script to produce the necessary files.
mt> >
mt> > Has somebody got these things and can mail me these
mt> files! I could not
mt> > found them in the mailing history.
mt> >
mt> > Has somebody also dealed with dbtext and can give me some example
mt> > files for ser.cfg and dbtext files for
mt> "authentication/subscriber" and
mt> > "location".
mt> >
mt> rhe readme file of dbtext module has some examples of table
mt> structures
mt> as well as a simple config file.
mt>
mt> http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/mod
mt> ules/dbtext/README?rev=HEAD&content-type=text/vnd.viewcvs-markup
mt>
mt> Daniel
mt>
mt> > Thanks in advance
mt> >
mt> > Thorsten Mueller
mt> >
mt> > ser 0.8.14 cross-compiled for arm
mt> > My ser.cfg:
mt> > ####################################################
mt> > ...
mt> > # Added by mt for authentication with dbtext
mt> > loadmodule "/usr/local/lib/ser/modules/auth.so"
mt> > loadmodule "/usr/local/lib/ser/modules/auth_db.so"
mt> > loadmodule "/usr/local/lib/ser/modules/dbtext.so"
mt> >
mt> > modparam("auth_db", "db_url","dbtext:///var/dbtext/ser")
mt> >
mt> > # -- auth params --
mt> > modparam("auth_db", "calculate_ha1", 1)
mt> > modparam("auth_db", "password_column", "password")
mt> > modparam("auth_db", "user_column", "username")
mt> > modparam("auth_db", "domain_column", "domain")
mt> > ...
mt> >
mt> > if (uri==myself) {
mt> >
mt> > if (method=="REGISTER") {
mt> >
mt> > if (!www_authorize("domain.com", "subscriber")) {
mt> > www_challenge("domain.com", "0");
mt> > break;
mt> > };
mt> > save("location");
mt> > break;
mt> > }
mt> > ...
mt> > #####################################################
mt> >
mt> > I have a file in /var/dbtext/ser which is called "subscriber":
mt> > #####################################################
mt> > username(str) password(str) ha1(str) domain(str) ha1b(str)
mt> > suser:supasswd:xxx:domain.com:xxx
mt> > #####################################################
mt> >
mt> >-----------------------------------------------------------
mt> -------------
mt> >
mt> >_______________________________________________
mt> >Serusers mailing list
mt> >serusers(a)lists.iptel.org
mt> >http://lists.iptel.org/mailman/listinfo/serusers
mt> >
mt> >
mt>
Hi serusers,
I have been using various versions of SER from last year without any problem
but recently I made a new installation of OpenSER 0.9.5. Since then I am
having problems with digest authentication from some of the phones. I have a
bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the
server, while all the soft phones can register successfully. The server says
Credentials with given realm not found. I tried to change the realm to
localhost, the IP address, with no luck.
And below is the result of ngrep
I tried to grep the messages from the phones and here below is one message
from a Cisco 186 ATA which has failed to register
########### Beginning of the capture ##################
U PHONEIP:5060 -> SERVERIP:5060
REGISTER sip:SERVERIP SIP/2.0.
Via: SIP/2.0/UDP PHONEIP:5060.
From: <sip:06090003@SERVERIP;user=phone>;tag=500808430.
To: <sip:06090003@SERVERIP;user=phone>.
Call-ID: 704382462@PHONEIP.
CSeq: 1 REGISTER.
Contact: <sip:06090003@PHONEIP:5060;user=phone;transport=udp>;expires=3600.
User-Agent: Cisco ATA 186 v2.16.2 ata18x (030829a).
Content-Length: 0.
#
U SERVERIP:5060 -> PHONEIP:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP PHONEIP:5060.
From: <sip:06090003@SERVERIP;user=phone>;tag=500808430.
To: <sip:06090003@SERVERIP;user=phone>.
Call-ID: 704382462@PHONEIP.
CSeq: 1 REGISTER.
Server: OpenSer (0.9.5 (i386/linux)).
Content-Length: 0.
Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490
req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP
out_uri=sip:SERVERIP via_cnt==1".
#
U SERVERIP:5060 -> PHONEIP:5060
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP PHONEIP:5060.
From: <sip:06090003@SERVERIP;user=phone>;tag=500808430.
To:
<sip:06090003@SERVERIP;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8af0
.
Call-ID: 704382462@PHONEIP.
CSeq: 1 REGISTER.
WWW-Authenticate: Digest realm="talk.artel.rw",
nonce="42edb29e1dbcc6fa814dd3396634ed7be68eea56".
Server: OpenSer (0.9.5 (i386/linux)).
Content-Length: 0.
Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490
req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP
out_uri=sip:SERVERIP via_cnt==1".
Any idea?
Aimable
Does anyone have info on adding Asterisk Support for Voice Messages, that page is missing from the SIP - Getting Started - Issue 04a.doc www.ONsip.org ?
Hi my friends!
I have been tested my SER config for a few weeks (ser-0.8.14). It's OK,
but in the ser's MySQL database in the acc table appears a few row with
sip_method=PRACK and sip_method=INFO (there are > 1100 rows with
INVITE,ACK and BYE). What is this meaning? I cannot regenerate this row.
can anybody help me?
Thanks
Szabolcs
Hi,
Can I use my existing radius server as my login authentication for
ser? The existing radius uses the system to read the user accounts, but
explained on the radius howto i must create the user accounts on users
file of the freeradius.
Please help.
Thanks,
--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
Could the 183 session progress have anything to do with the one way
voice ?
________________________________
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: Monday, August 01, 2005 4:54 PM
To: Sam Lee; serusers(a)lists.iptel.org
Subject: Re: [Serusers] 1 way voice issue
Very hard to say without a complete ngrep trace. If both parties are on
public IPs, there is probably something with the signalling, not the SDP
payload. The 183 Session Progress is a so-called provisional reponse.
It can be used for early media (to set up the rtp stream before
connection). A double 183 sounds strange if it's the proxy that does
it.
g-)
---- Original Message ----
From: Sam Lee
To: serusers(a)lists.iptel.org
Sent: Monday, July 25, 2005 12:27 PM
Subject: [Serusers] 1 way voice issue
> Hi,
>
> I've recently encountered some problem with my SIP service whereby i
> call out to a specific number and i encounter a one way voice. If i'm
> the initiator, i cannot hear the other party but he can hear me. At
> first i thought it was a return route issue (as i'm going thru NAT) ,
> so i switch my SIP to a public IP but i still face the same problem.
> Its really only that specific PSTN number that i have dialed facing
> this problem. The only difference that i can think of is that PSTN
> number is on a different route. I did a NGREP from my SIP server for
> the PSTN number that works (2-way voice) and the Number that doesn't
> work (1-way voice) . The only difference is there is an extra :
>
> NGW --> Proxy
> SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP
> Proxy --> SIP Device
> SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP
>
>
> for the PSTN number that works (the one with 2-way voice).
>
> Anyone has idea what does the Session Progress is for ? Or what
> problem am i facing ?
>
> Thanks a mILLION !
>
> Regards,
> Sam
>
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
hi there,
Can you please tell me if I can use the permissions module to trust the requests coming towards my SIP server using source IP. But my problem is I want to use flat files to store the source IPs I trust. Is it possible
And if the IPs are trusted I need to forward them without any authentication.
I need to store the trusted IPs in flat files since I dont need the overhead of querying DB.
Please help me out.
Thanx
---------------------------------
Start your day with Yahoo! - make it your home page
Hi List,
I wonder if you could share your experience with me of running SER behind NAT. Are there any known problems with doing it this way? - Server side NAT and have SER fix NAT'ed clients and (Media)Proxying them where needed.
My specific setup is as follows:-
SER(0.9.13) with a static private IP
Firewall performing static on-to-one NAT
Users will be on public IP's/(behind NAT)
Does this cause any problems with the SIP/SDP packets?
Also it is the intention to run Asterisk in the same way to provide voicemail, will this setup produce any foreseeable problems?
Many Thanks,
Alan