Hi,
Has any one done using Asterisk/Digium T1 card as PSTN gateway, and SER as
the backend soft switch?
I am looking to install Asterisk boxes in a few cities to act as PSTN
gateways and using SER to accept these calls. SER will have to communicate
with other long distance carriers to allow world wide calling.
Will this combination work? I like to know your comments or suggestions on
this combination.
thanks
Tony Lum
Problems:
1. SER do not start properly after reboot.
2. Can´t not log on to SIP-server trough NAT.
Description:
1. SER do not start properly after reboot.
First of all i must point out that i have no more than a couple of weeks
experience. I downloaded the "Admin guide" and the "Dan Austin howto" as
reference dokumentation to set up a basic SER server. I followed it step
by step and tested it. I found out that ive got some problem with ser when
i restarted the machine. I restarted the machine, startted ser with "ser
start" witch gived me:
Listening on:
127.0.0.1[127.0.0.1]:5060
192.168.1.3[192.168.1.3]:5060
But when i was checking the status with "serctl monitor" It semed to me as
ether the ser server or the ser_fifo was down cause it dident give me any
status att all.
when i checked the lokal processes with "ps aux" it dident give me
anything on "ser start" as it usualy do. So i tried out the default
ser.cfg and it worked.... ps aux gived me "ser start". Then i tried the
default ser.cfg with mysql auth support and it worked. I rebooted and the
same problem all over again.
2. Can not log on to SIP-server trough NAT.
I can make local connections and initiate calls but not trough my
dial.mine.nu adress. I have routed port 5060-5062 in my firewall against
the server machine.(i think it would be sufficient to at least log on) I
can make connections against iptel.org reference server but not against my
local server.
Enviromental variables:
OS: Slackware 9.1 (Linux)
client: Kphone and W-messenger.
server: ser 0.8.12
Host-name:SIPserver.hasselan
Domain: hasselan
SIP-domain: SIPserver.hasselan
Mysql: 4.0.18
external domain: dial.mine.nu
configurationsfile= default ser.cfg + auth modules and mysql support.
following lines were edited:
www_authorise =SIPserver.hasselan
www_challenge SIPserver.hasselan
hello friends,
i have installed free radius server-0.9.3 and radius
client-0.3.2
and followed the ser howto
radtest is success full
and i updated dictionary of radius client with of in
web
available dictionary.ser ( of sip related attributies>
and i included statement of INCLUDE <path of
dicionary.ser>
when i start radiusd -x its givens error as
Errors reading dictionary: dict_init:
/usr/local/etc/raddb/dictionary[23]: Couldn't open
dictionary "
/usr/local/etc/raddb/dictionary.ser": No such file or
directory
Errors reading radiusd.conf
so i gone to "/usr/local/share/freeradius/"
path and appended whole dictionary.ser contets into it
and also kept link as INclude dictinary.ser
even though if i create packet "digest"
User-Name = "test", Digest-Response =
"631d6d73147add2f9e437f59bbc3aeb7",
Digest-Realm = "testrealm", Digest-Nonce = "1234abcd"
,
Digest-Method = "INVITE", Digest-URI =
"sip:5555551212@example.com",
Digest-Algorithm = "MD5", Digest-User-Name = "test"
and run with
root@/usr/local/src# radclient -f digest localhost
auth <shared_secret>
it s giveng error as
radclient:No token read where we expected an attribute
name
i checked both in client and server
dictionary attributes are present for the all that
which are included in the pakcet.
if include only
User-Name = "test",User-Password="test" in the digest
packet
and check it s sucessfull so what may be the wrong
sip method digest packet
please help me
with regards
rama kanth
__________________________________
Do you Yahoo!?
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Again: Always send a copy to the mailing list so that others can see the
replies, do not send pure private emails !
Regarding to the permission problem, see
http://lists.iptel.org/pipermail/serusers/2003-December/004607.html
Jan.
On 25-05 16:16, sidf@blrcsv wrote:
> Thanks. The trouble was resolved successfully through your suggestion.
> But i met another problem:
>
> *****On user interface site (logged in) ***************
> Warning: fopen("/tmp/ser_fifo", "w") - Permission denied in
> /var/www/html/functions.php on line 206
>
> sorry -- cannot open write fifo
>
> ************************************************************
> Thanks again.
>
>
> ----- Original Message -----
> From: "Jan Janak" <jan(a)iptel.org>
> To: "sidf@blrcsv" <sidf(a)blrcsv.china.bell-labs.com>
> Cc: <serusers(a)lists.iptel.org>
> Sent: Tuesday, May 25, 2004 4:08 PM
> Subject: Re: [Serusers] SERWEB: How i change the confirmation address in the
> confirmation email to user?
>
>
> > On 25-05 15:11, sidf@blrcsv wrote:
> > > I installed the SERWEB0.8.12. After i register a new user using the
> register webpage, a confirmation email will be sent to the user's email
> address. it look like this:
> > >
> > > Thank you for registering with ......
> > >
> > > We are reserving the following SIP address for you:
> > > sip:8002@......
> > >
> > > To finalize your registration please check the following URL within 24
> hours:
> > >
> http://....../serweb/user/reg/confirmation.php?nr=bd77f30ed0c805c6d204e65eae
> 5aa9d5
> > >
> > > (If you confirm later you will have to re-register.)
> > > My question is , which and how modify the php file in directory
> /serweb/user/reg , can make me change the registration confirmation
> address?Thanks.And , I find a bug in SERWEB, default address in the
> confirmation email is ...../serweb/user/... ,but it is
> serweb/user_interfacein the serweb package.siduanfeng
> >
> > What exactly do you want to change, I am not sure I understand ? In
> > regards to the user_interface problem, we usually create an alias
> > user->user_interface in the configuration of apache.
> >
> >
> > Jan.
>
I have a problem with redirection to voice mail. If caller hangs up,
the CANCEL hits SER and the call is cleanly terminated. SER however
continues to failure_route after timeout of the initial INVITE which
results in an empty message generated by the voice mail server.
Does anyone know how to deal with this?
Thanks,
Adrian
if (is_user_in("Request-URI", "voicemail")) {
setflag(3);
};
if (!lookup("location")) {
xlog ("L_INFO","Location not found");
} else {
if (method == "INVITE" && isflagset(3)) {
xlog ("L_INFO","Flag3 is set, try voicemail after timeout");
t_on_failure("3");
};
};
xlog ("L_INFO","Relay to destination");
xlog ("L_INFO", "%rm from %is: %fu -> %ru\n");
t_relay();
--------
failure_route[3] {
revert_uri();
xlog ("L_INFO","Forward to voicemail (failure route 3)");
exec_dset("/etc/ser/serredir.py voip_voicemail ;echo>/dev/null");
append_branch();
t_relay();
}
Hello,
Is there anyone in the list who managed to use AG's mediaproxy with stable version of ser?
I tried following their instructions ie get module code from cvs and compile in stable source tree.
It compiles but ser crashes in 'del_lump' function whenever any mediaproxy's function is called.
Thanks,
Alex
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi Group,
Is anyone running prepaid service on his ser ?
and can he/she give me some ideas how to do it ;-)
Thanks :)
>On Monday 31 May 2004 11:07, you wrote:
> > I'm using not just normal voicemail, but IVR system who can be verry
> > helpful for you...
>
> kamen,
>
> is it now possible to listen voicemails also by phone using the ivr
> system or is email still the only option?
>
> -- juha
- --
Regards, Kamen Sharlandjiev
System Administrator
NetBG Communication
Tel: +359 2 962 43 52
+359 2 962 53 93
- --
Public GPG key at: http://pgp.mit.edu
pub 1024D/C6347D3D 2003-03-19 Kamen Sharlandjiev (Comment) <stone(a)netbg.com>
Key fingerprint = D6AA 55BD F0CB FC6B 8C22 A13B 91D1 55C3 C634 7D3D
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Version: GnuPG v1.2.3 (GNU/Linux)
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yjpVPxlHcrFkq6ifYgtOO1s=
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Dear All,
I'm a long-term Asterisk user and have two data centers using Asterisks to
send and receive calls. This works great and I service
I needed a capable SIP proxy and following some good reviews of SER I put a
copy of SER in my test lab.
Quad T1's<->Asterisk (PBX)<->(LAN<->DMZ)<->SER<->(Firewall)<->(Internet)
|
Local US Help Desk (Snom 200')
This setup works well. I can pass calls from over the internet to the
Asterisk PBX via SER using X-Ten Lit.
I have now loaded Ser WEB and like the functionality. I understand to use
the accounting, missed calls and voicemail I need to use the sem modules?
I downloaded the sem modules from the cvs repository on belios.de.
I can't really find any documentation on how to integrate voicemail.
At the moment if I don't have one of my users registered with ser and I try
to dial them I get 404 error.
I would really appreciate some help;
1. How do I make the system emulate the phone ringing and not just get a
404?
2. How do I send the call to Voicemail after 10 rings if the user is not
available?
Warm Regards
Shad Mortazavi
-----------------------
Nexus Technical Manager
n|m Nexus Management Inc
Sydney
hi my name is takuya
i'm japanease
i wanna install ser to FreeBSD
but i can't!!
becase i getting error, like this
master# make
"Makefile.sources", line 17: Missing dependency operator
"Makefile.sources", line 19: Need an operator
Error expanding embedded variable.
i seraching reason,but i can't find
please someone teach me.
--------------------------------
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TEL 03-5297-8011
Mail:imaizumi(a)ap-com.co.jp
--------------------------------