HeLLO ,
Recently i have installed SER 0.9.3 -0.2 in my system for
testing purpose. I would like to install serweb too in my system but
when I scaned thru the webpage i couldnt find the way to install and the
complete documentation.So can anyone help me in this regard.
Thank u
BR
S.Kumar
Hello, i have to transform the dialed digits like
this:
0[2-3][2-9][2-9]15[4-6]XXXXXX to 549[2-3][4-6]XXXXXX
for example: 0385 15 6097775 to 549 385 6097775
I can do it with prefix and strip but only by adding
lots of config lines.
Is there any other easy method to do it?
Is it possible to use a translation table in mysql to
do the trick?
regards, Pablo.
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I've trying to add support to the dispatcher module for reloading its
configuration file on a fifo command. This allows an external daemon to
monitor a set of external Asterisk machines, detect when they fail, and
round robin calls between the surviving ones.
I'm (I think) most of the way there. I can issue the command, and see
the reload happening, but I see a core dump:
[root@test dispatcher]# serctl fifo dispatcher_reload
2(31672) DISPATCHER:ds_load_list: dest [135343048/135343048/21]
<sip:1.2.3.4:5060>
2(31672) DISPATCHER:ds_load_list: dest [0/1/1] <sip:1.2.3.4:5060>
2(31672) DISPATCHER:ds_load_list: found [1] dest sets
0(31670) child process 31672 exited by a signal 11
0(31670) core was generated
(IP address changed to protect the guilty)
I've had a look at the code, and don't really understand why it's
happening. A gdb backtrace shows:
(gdb) bt
#0 ds_load_list (lfile=0x8112b98 "È+\021\b\025") at dispatch.c:281
281 dp = dp->next;
#0 ds_load_list (lfile=0x8112b98 "È+\021\b\025") at dispatch.c:281
#1 0x0017cacf in dispatcher_reload (pipe=0xa1f5d30,
response_file=0x8112378 "/tmp/ser_receiver_31623")
at dispatcher.c:170
#2 0x08057456 in start_fifo_server () at fifo_server.c:540
#3 0x0805caf5 in main_loop () at main.c:988
#4 0x0805e52b in main (argc=3, argv=0xbffa3584) at main.c:1568
I've modified dispatcher.c. In mod_init, I've added:
if (register_fifo_cmd(dispatcher_reload, "dispatcher_reload",
0) < 0)
{
LOG(L_ERR, "Cannot register dispatcher_reload\n");
return -1;
}
and added a new function:
static int dispatcher_reload ( FILE* pipe, char* response_file )
{
ds_destroy_list ();
if (ds_load_list(dslistfile)==0) {
fifo_reply (response_file, "200 OK\n");
return 1;
} else {
fifo_reply (response_file, "400 Dispatcher reload
failed\n");
return -1;
}
}
Can anyone shed light on this?
--
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
I need a way to store somewhere some computed values relative to sip
messages. That is, I need some kind of "variables"!
I need to access to these "variables" both from a custom SER module and
from the ser.cfg script.
I think that the solution are AVPs.
Am I right?
But I'm a little confused about the many "incarnations" of the AVP.
After looking a little around I have understood that from inside the
module I have to use the functions in "usr_avp.c". Am I right?
Then there are the "avp" and "avpops" modules.
My question is: they all refer to the SAME avps?
Thas is, if I save an avp value by means of "usr_avp.c", then I can
access the SAME avp by means of "avpops"?
Or they are different and separate implementations?
Thanks.
P.S.
I'm using SER 0.9.3.
--
___________________________________________________
__
|- giannici(a)neomedia.it
|ederico Giannici http://www.neomedia.it
___________________________________________________
Hi
Is there any standard on call timeouts. I hit a really strange problem.
I have the following set
PSTN ---> SER ---> asterisk
All ip devices register to SER
Now when a call comes in via PSTN and it destined for a coporate, it
hits asterisk, which give its the auto attendant features and all, press
1, and you goto sales, via ser. Now if no one at sales answers, it goes
to voicemail in asterisk again.
BUT when I make a call from within that company, say to extension 111,
which dials out to a IP phone registered in SER, and no one answers, I
dont trip to voicemail.
After hrs of testing, and a some help from some unknowns on the web, I
realised it seemed to be a timeout problem. The call in SER was timing
out, and hence call was being destroyed, if I set a timeout in the Dial
string in asterisk to a number (which just by luck happened to be less
than that in ser) it all worked.
Was just wondering if there were any standard timeout parameters which
people used for call, cause calls to pstn gateways would take longer
than sip to sip ones.
Iqbal
Oznan,
CDRTool has a rating and prepaid engine suitable for SER, Asterisk and
Cisco. It is a commercial product.
See: http://cdrtool.ag-projects.com
Adrian
>>>>>>>>>>>>>>
Dear List,
As some of you know, there would be an open-source-billing for OpenSER,
i can see call duration,add remove edit users etc features but missing
thing is rating. Since there may be more then one rating table it's
very hard for MySql to query whole database and gives time out errors.
I can manage PHP to create/delete/edit a new database with
prefix,location name,initial time,initial cost,increment time,increment
cost and rate id. Now there's a problem, think that there're 2
different rate table for different customers, a rate id is what i would
like to write to acc table in each call. May be i should force ACC
module to add rate_id together with other information like from_uri,
to_sip, timestamp etc. etc. How can i manage this, is there any clue or
information ?
Thanks,
Ozan Blotter
Hello,
I posted
http://lists.iptel.org/pipermail/semsdev/2005-August/000437.html
Nobody is interested to improve sems features !
It might not be the best solution.
Some people are stranges when somebody suggest
features
nobody reply.
therefore we can read sometimes some developpers
telling "users just wait we provide a good and free
project".
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Dear List,
As some of you know, there would be an open-source-billing for OpenSER, i can see call duration,add remove edit users etc features but missing thing is rating. Since there may be more then one rating table it's very hard for MySql to query whole database and gives time out errors. I can manage PHP to create/delete/edit a new database with prefix,location name,initial time,initial cost,increment time,increment cost and rate id. Now there's a problem, think that there're 2 different rate table for different customers, a rate id is what i would like to write to acc table in each call. May be i should force ACC module to add rate_id together with other information like from_uri, to_sip, timestamp etc. etc. How can i manage this, is there any clue or information ?
Thanks,
Ozan Blotter
Dear List,
As some of you know, there would be an open-source-billing for OpenSER, i can see call duration,add remove edit users etc features but missing thing is rating. Since there may be more then one rating table it's very hard for MySql to query whole database and gives time out errors. I can manage PHP to create/delete/edit a new database with prefix,location name,initial time,initial cost,increment time,increment cost and rate id. Now there's a problem, think that there're 2 different rate table for different customers, a rate id is what i would like to write to acc table in each call. May be i should force ACC module to add rate_id together with other information like from_uri, to_sip, timestamp etc. etc. How can i manage this, is there any clue or information ?
Thanks,
Ozan Blotter
