Hello ser users,
I've sent many mails about starting ser.
my problem is solved .
ser sent to syslog:
auth_db:str_fixup: Invalid table version (use
ser_mysql.sh reinstall)
auth_db module did not find table "subsciber" below.
if (!www_authorize("nxs.yi.org", "subsciber"))
{
www_challenge("nxs.yi.org", "0");
sl_send_reply("401", "Unauthorized");
break;
};
I fix table to "subscriber" !
Thanks ser users for help
Regards
Harry
_____________________________________________________________________________
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I am trying to route a call from a non-routable 10.x.x.x network to a
phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to
SIP from my asterisk to an x-lite
I found some references indicating that I should call:
if ( src_ip = IP_GATEWAY)
{
force_rport();
fix_contact();
use_media_proxy();
};
Supposedly, you need the force_rport and the fix_contact to put media
proxy in the middle of the two rtp sessions, but I have et to see this
actually work. Has anyone been able to pass a call through media proxy
from a GW on a non-routable to a routable network, or even between two
ip's on the same network as the mediaproxy?
If so, please share your configs.
-----Original Message-----
From: bogdan aux [mailto:aux1d@yahoo.com]
Sent: Friday, May 13, 2005 6:16 AM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Mediaproxy in LAN (?)
I want to route RTP packets between 2 computers in the
same network through Mediaproxy. I know this may have
no sense, but it's just a test configuration.
The call use_media_proxy() seems to have no effect,
the RTP packets go directly from one computer to
another.
My guess is mediaproxy makes a test on the IPs (they
are 192.168.147.2 and 192.168.147.3) and refuses to
proxy the call.
Is mediaproxy working just between different networks
or it's an error in my configuration file?
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Is anyone using the CDRTool from AG I have installed and configured it for
testing. Its missing the Rating engine etc.
Anyone know the cost of a Lic with the rating engine ?
I read about SER and I have installed Asterisk on one machine.
Unfortunately this machine is broken and the replacement will take
longer as I wished.
I have a P-4, 1GHz with 1 GB RAM, which was planned to be used as SER
anyway. I would like to use this downtime of the Asterisk box to install
SER.
I understand that SER and Asterisk can work together, while SER makes
connection, Asterisk will add features.
How do I do that efficiently?
1. SER needs a public IP, does then Asterisk need a pulic IP, or can it
just be on the internal LAN?
2. Now the password for each phone is made in the SIP and IAX config
files. How can I do that than in SER? If I use reatime, can I use the
same database?
3. A user can make a phone call to the users on our system (Asterisk
domain) and via gateways. We use ASTCC for billing. How can I use this
together with SER.
I am not sure if I understand it correct, but would for internal calls
(calls without features) be handeled directly from SER, while if the
user need a feature, than he would be transfered to Asterisk? How can I
setup multiple Asterisk boxes with one SER, but all Asterisk machines
share the same dial plan (realtime)
I wanted to read more first, but the down time of my * forces me to do a
SER setup faster, so that my users can quickly (by office hours Monday)
at least make internal calls.
Any hint is welcome ;-)
bye
Ronald
Dear Friends,
I know this is annoying to post nearly same things but still no answer :( Is
it really so hard or can anybody tell me why this is not working.
There are :
ATA 1 : 12345 (registered and can call 54321, no problem)
ATA 2 : 54321 (registered and can call 12345, no problem)
ANALOG FXO : 201 (registered as 201, for test purposes i only registered one
port of it)
With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it worked,
even i can add more prefixes no problem. But in SER i cannot :( I'm using
ver 0.9.2. Normally it's like when i want to call a GSM, proxy simply adds a
prefix 201 which is same with registered sip number for regarding analog
port and forwards it (looks like sip:20105353490056@192.168.1.35) to Gateway
and Gateway simply strips 201 from it and dials 05353490056, yes it
connects. But what this same does not work on SER, any ideas because it's
very boring to play all day and night ?
Thanks....
>>>>>>>>>> minimal SER.CFG >>>>>>>>>>
debug=3
fork=yes
log_stderror=no
check_via=no
dns=no
rev_dns=no
listen=192.168.1.10
port=5060
children=4
fifo="/tmp/ser_fifo"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
route {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
if (uri=~"^sip:053[0-9]*@.*") {
prefix("201");
rewritehostport ("192.168.1.35:5060");
# forward ("192.168.1.35:5060");
t_relay_to_udp("192.168.1.35","5060");
break;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
>>>>>>>>>> call from 12345 to 54321 (SIP 2 SIP Call) >>>>>>>>>>
U 192.168.1.10:5060 -> 192.168.1.201:5060
INVITE sip:54321@192.168.1.201:5060 SIP/2.0..Record-Route:
<sip:192.168.1.1
0;ftag=xITO2cDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9hG4bK233f
.8f8b9d41.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bKba4b3e1d3eed
9c7c..Max-Forwards: 16..To: <sip:54321@192.168.1.10>..From:
<sip:12345@192.
168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID:
911D1B912D91212(a)192.168.1.20
0..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..Session-Expires
: 100;refresher=uac..Supported: timer..Content-Type:
application/sdp..Conte
nt-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-..c=IN IP4
192.
168.1.200..t=0 0..m=audio 2142 RTP/AVP 18 18..a=rtpmap:18
G729/8000..a=send
recv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bKba4b
3e1d3eed9c7c..To: <sip:54321@192.168.1.10>;tag=xETOxUDMxED..From:
<sip:1234
5(a)192.168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID:
911D1B912D91212(a)192.16
8.1.200..CSeq: 1 INVITE..Record-Route:
<sip:192.168.1.10;ftag=xITO2cDMxID;l
r=on>..Content-Length: 0....
>>>>>>>>>> call from 12345 to 05353490056 (GSM Call) >>>>>>>>>>
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
192.16
8.1.200:5060;branch=z9hG4bK31bc676961f91e18..To:
<sip:05353490056@192.168.1
.10>..From: <sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
8
76D1B912D91212@192.168.1.200..CSeq: 1 INVITE..Server: Sip EXpress router
(0
.9.2 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.10:5060
"Nois
y feedback tells: pid=4587 req_src_ip=192.168.1.200 req_src_port=5060
in_u
ri=sip:05353490056@192.168.1.10
out_uri=sip:20105353490056@192.168.1.35:506
0 via_cnt==1"....
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 200 ok -- no more pending branches..Via: SIP/2.0/UDP
192.168.1.200:
5060;branch=z9hG4bK31bc676961f91e18..To:
<sip:05353490056@192.168.1.10>;tag
=2f9bfc2acf470ceacf4efdebbaa289b4-026c..From:
<sip:12345@192.168.1.10;user=
phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D91212@192.168.1.200..CSeq: 1
CAN
CEL..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length:
0..Wa
rning: 392 192.168.1.10:5060 "Noisy feedback tells: pid=4586
req_src_ip=19
2.168.1.200 req_src_port=5060 in_uri=sip:05353490056@192.168.1.10
out_uri=s
ip:20105353490056@192.168.1.35:5060 via_cnt==1"....
I want SER to use another application's database but can't see a way to tell
the auth_db module which table to use. I can see the paramaters for the
host/database name, and the column names. But how do I tell SER which table
to use or does it only ever use subscriber?
Regards
Cameron
Hi everybody,
I want to test my billing system based on the CDR data I've recorded at ACC table, but there is a problem, the data is not being recorded as I thought. I don't know why, when I go directly through mysql and make a "select * from acc;" it gives me:
Empty set (0.00 sec)
I don't know why the CDR date is not being recorded, I looked for it at the forum and google, and found something about mysql installation. It seems that I haven't compiled mysql with ACC support. Is that right ? Did someone have the same problem already ?
Thanks in Advance
Best Regards
--
Felipe Martins
Mundivox Communications
Tecnologia e Projetos
fmartins(a)mundivox.com
Tel.: +55 +21 +3820 8839
Cel.: +55 +21 +9823 8602
Fax.: +55 +21 +3820 8844
www.mundivox.com
Hello,
is it possible to use avp_write on the responses in failure route?
If yes, Contact header field could be evaluated and pushed into RURI,
then routing via route(x) to a new destination and finally solve the 3xx
forwarding issue.
Regards,
Martin
Hi
Does anyone have a dump of what a succesful call transfer for the
following two scenarios looks like
PSTN <---> PHONE A
PHONE A -----> PHONE B
PSTN <----> PHONE B
basically A transfer the call to B
the next scenario is the same but instead of PSTN , its between IP only
PHONE A <---> PHONE B
PHONE B ---> PHONE C
PHONE A <---> PHONE C
tks
Iqbal
hi guys,
is it possible to use a global phone book with ser?
i found no features like this in sip rfcs, but this is imho very
impotant thing for companys etc.
do anybody know an way to implement a global phone book for all users
of one ser?
bg,
grigory
