Assuming you relay properly, when Asterisk asterisk hangs up it should
pass the BYE to SER then SER relays it to the next party(s). This
doesn't sound like an acc problem, as you might have guessed :-)
I would have to see most of your config. When you say you have to "click
hangup" what do you mean.. on xlite? Is xlite the UA or is it hanging
off of Asterisk? If not Asterisk, what's hanging off of Asterisk?
Matt
-----Original Message-----
From: Barry Murphy [mailto:barry@unix.co.nz]
Sent: Tuesday, April 26, 2005 7:17 PM
To: Barry Murphy; Matt Schulte
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] ACC into mysql
Sorry for replying to my own post, however I have discovered
something.
When I call asterisk and asterisk hangup, my UA is still
connected till I hangup myself, however SER does see the hangup as I
looked in the acc serweb.
<sip:03@voip.fast.co.nz> <javascript:
open_ctd_win2('sip%3A03%40voip.fast.co.nz',
'sip%3A6494485566%40voip.fast.co.nz');> today 12:09
00:00:29 caller
<sip:03@voip.fast.co.nz> <javascript:
open_ctd_win2('sip%3A03%40voip.fast.co.nz',
'sip%3A6494485566%40voip.fast.co.nz');> today 12:09
00:00:50 callee
So my question is, if the remote end hangs up, how do I
disconnect the caller end?
Using XLite.
Barry
----- Original Message -----
From: Barry Murphy <mailto:barry@unix.co.nz>
To: Matt Schulte <mailto:mschulte@netlogic.net>
Cc: serusers(a)lists.iptel.org
Sent: Wednesday, April 27, 2005 12:13 PM
Subject: Re: [Serusers] ACC into mysql
Thanks,
After months of trying to get this working, your
solution fixed the problem. Now the only issue I have is the following:
When I call 03 ( UA -> SER -> Asterisk ) the date and
time is read out to the user, asterisk then hangs up the call, however
SER doesn't see the hangup, I have to manualy click hangup and then SER
gives the BYE.
Apr 27 12:09:04 max /usr/local/sbin/ser[30538]: ACC:
transaction answered:
call_id=93EF8C43-C9E2-4E4C-997C-B1A4DA8FD137(a)10.200.3.173,
totag=as113261f7, from=6494485566
<sip:6494485566@voip.fast.co.nz>;tag=3875371437,
i-uri=sip:03@voip.fast.co.nz, method=INVITE,
o-uri=sip:03@202.150.105.150:5070, fromtag=3875371437, code=200,
to=<sip:03@voip.fast.co.nz>;tag=as113261f7, uid=n/a, userpart=03
Apr 27 12:09:04 max /usr/local/sbin/ser[30541]: ACC:
request acknowledged:
call_id=93EF8C43-C9E2-4E4C-997C-B1A4DA8FD137(a)10.200.3.173,
totag=as113261f7, from=6494485566
<sip:6494485566@voip.fast.co.nz>;tag=3875371437,
i-uri=sip:03@202.150.105.150:5070, method=ACK,
o-uri=sip:03@202.150.105.150:5070, fromtag=3875371437, code=200,
to=<sip:03@voip.fast.co.nz>;tag=as113261f7, uid=n/a, userpart=03
Apr 27 12:09:33 max /usr/local/sbin/ser[30537]: ACC:
transaction answered:
call_id=93EF8C43-C9E2-4E4C-997C-B1A4DA8FD137(a)10.200.3.173,
totag=as113261f7, from=6494485566
<sip:6494485566@voip.fast.co.nz>;tag=3875371437,
i-uri=sip:03@202.150.105.150:5070, method=BYE,
o-uri=sip:03@202.150.105.150:5070, fromtag=3875371437, code=200,
to=<sip:03@voip.fast.co.nz>;tag=as113261f7, uid=n/a, userpart=03
Any ideas?
Thanks
Barry
----- Original Message -----
From: Matt Schulte
<mailto:mschulte@netlogic.net>
To: Edgardo O. Gonzales II
<mailto:edgardo.g@pacific.net.ph> ; Kofi Obiri-Yeboah
<mailto:kofi@radiocomplex.com>
Cc: serusers(a)lists.iptel.org
Sent: Wednesday, April 27, 2005 1:13 AM
Subject: RE: [Serusers] ACC into mysql
This particular config (I don't think?) won't
log to a DB, additionally you need to setflag(1); somewhere in your
config. We tried to set on "outbound" only originally and didn't have
much luck so I set it near the top. To log to (my)sql you will
additionally need to set another flag and setup similar mod params:
modparam("acc", "db_url",
"mysql://ser:serro@blah.mysql.haha/dbname")
# Note flag 2, you will need to setflag(1); AND
setflag(2);
modparam("acc", "db_flag", 2)
#Note this one logs all failed calls from the
invite response, I find it useful
modparam("acc", "failed_transactions", 1)
# In your route config use something like the
following
# This is pretty much near the top, while ACC
won't log reg's or INFO anyway, I just thought it'd be a little cleaner
to have
...
if (!method == "REGISTER" || !method == "INFO" )
{
setflag(1);
setflag(2);
}
...
# Doing the above ensure everything gets tagged
including BYE's. A good reason why your BYE's may not get
# tagged is possibly because record-route could
be relaying the call before the flag gets set, just a thought..
_____
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serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi all,
I am trying to design a VoIP solution covering multiple prefixes
(cities) and providing SIP-SIP, SIP-PSTN and PSTN-SIP calls.
The alternatives I am considering are:
- a single SER installation to register SIP clients (that would be PC
softphones mostly) and multiple asterisk servers in each city, and
then have SER route calls directly to the appropriate asterisk for
PSTN termination;
- central SER should forward all PSTN-destined calls to the central
asterisk(residing on the same PC), which would then rforward it to the
appropriate remote asterisk via IAX. I think this would be more
suitable for billing, since all billling should be done by the
asterisks;
- The other option I'm considering is installing multiple SER+asterisk
machines in each remote location, have local users register to the
appropriate SER and the forward calls between asterisks via IAX;
Which of these would you find most suitable? I think this would
interest other people as well.
Another thing I find interesting is bandwidth planning - I am in doubt
whether to deploy RTP Proxy in my initial design, since that would
increase bandwidth requirements dramatically. AFAIK SER only handles
signalling messages while most of the bandiwdth (RTP stream) would go
directly between the two endpoints without traversing my ISP link.
This way, only calls to and from the PSTN should burden my ISP link.
Am I right on that?
Dear all,
Please help I have problem rewriting the e.g"o=2093 8000 0 IN IP4 172.16.3.31.." the IP4 ip address to the public IP. Please advice what can be done. Thanks.
regards,
nicky
I'm trying to implement My acccount , with ser_0.8.14 and sems_2004-07-27 on the
same machine. All servers seem to be running alright, but when I dial a
user who's Myaccount option is enabled,
i am getting Sorry cannot open write fifo error.
How to solve the problem
pls mail me
The logs in /var/log/messages says
Apr 27 13:55:11 yourlocalphonenumber kernel: audit(1114581311.290:0): avc: denied { getattr } for pid=22807 exe=/usr/sbin/httpd path=/var/www/phplib/of_textarea.inc dev=dm-0 ino=16875942 scontext=root:system_r:httpd_t tcontext=root:object_r:var_t tclass=file
Apr 27 13:55:11 yourlocalphonenumber kernel: audit(1114581311.291:0): avc: denied { getattr } for pid=22807 exe=/usr/sbin/httpd path=/var/www/phplib/of_file.inc dev=dm-0 ino=16875960 scontext=root:system_r:httpd_t tcontext=root:object_r:var_t tclass=file
Apr 27 13:55:11 yourlocalphonenumber kernel: audit(1114581311.291:0): avc: denied { getattr } for pid=22807 exe=/usr/sbin/httpd path=/var/www/phplib/of_calendar.inc dev=dm-0 ino=16875932 scontext=root:system_r:httpd_t tcontext=root:object_r:var_t tclass=file
I know how to use avp_check to determine if a variable is defined but
I need to check if the same variable has a value or is null. Actually I
simply
want to check if it is not null. Is there a way to do this?
Thanks,Steve
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
Hello,
I just try to get ser up and running (version 0.9.0)
I didn't touch anything, just did a
make all ; make install
but get the following when trying to start:
>---snip---
[root@batman ser-0.9.0]# 0(0) Maxfwd module- initializing
textops - initializing
ERROR: error -478 while trying to fix configuration
<---snip---
Can anybody give me some hints?
Regards
Jochen
Hey guys,
Anyone know of a rating engine for ser? I got all my records in the
database, but now i need to match them up against voipjets charges and send
a bill to the users.
Any ideas?
Thanks
Barry
hi guys,
i have a query on sip.
I build a voice conference server using sip stack.The
server crashes if the number of users increses.How
can i fix this problem.
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hi guys,
when the ack will be in different transaction in sip?
________________________________________________________________________
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