Hi all,
I have installed CDRtool version 6.9.9. *I have set the Prepaid_lock to
1* to disable multiple calls from a single prepaid account. But the parallel
call gets established. How can I solve this problem
$RatingEngine=array(socketIP = xxx.xxx.xxx.xxx,
Bogdan,
I started the Opensips with debug=6 and there weren't changes on the
initialization log, I've used the strace (follow trace attached), I'm not an
expert in the strace, but the only thing that i could see was a message
14752 read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83
nbsp;Hi,
I want to know whether the opensips support SBC
with dual NICs,
If yes, whether opensips can be used
to construct a sbc cluster with the function of load
balancing,
and is there any performance test data as a
sbc?
ths,
yinlin___
Users
Ahmed,
One way to avoid the problem of Asterisk returning 491 Request Pending would be
to authenticate the
call between the UA and Opensips (see proxy_authorize and proxy_challenge
functions in modules auth
and auth_db) then relay it to Asterisk with IP address authentication only.
i.e. so
Hi Brian,
the error you posted has nothing to do with blocking / non-blocking -
it simply shows that opensips fails to open a new TCP connection because
there was nobody listening on the other side.
Regards,
Bogdan
opensipsl...@encambio.com wrote:
Hello list,
It seems that in August 2008
Hi Ha`,
There is a very simple example in the documentation:
route[b2b_request] {
xlog(b2b_request ($ci)\n);
}
route[b2b_reply] {
xlog(b2b_reply ($ci)\n);
}
You can call in these routes any function that you call in a request route.
Regards,
--
Anca Vamanu
www.voice-system.ro
ha do
Hi Brian,
There is a misunderstanding from your side in what the b2b scenario
documents are concerned ( please read carefully the documentation -
http://www.opensips.org/Resources/B2buaTutorial ). The important thing
is that there should only be rules in the scenario for requests that
need a
Hi Jan,
are you sure you have the opensips-mysql-module_1.6.1.0_amd64.deb file
in the current dir (or name is correct) ??
I just tried the install of the same packages (freshly generated from
svn tree) and I had no issue.
Regards,
Bogdan
Jan D. wrote:
Bogdan,
Thanks for the quick response,
Hi Antonio,
The relevant part is:
14752 socket(PF_FILE, SOCK_STREAM, 0) = 5
14752 fcntl(5, F_SETFL, O_RDONLY) = 0
14752 fcntl(5, F_GETFL) = 0x2 (flags O_RDWR)
14752 connect(5, {sa_family=AF_FILE,
path=/var/run/mysqld/mysqld.sock}, 110) = 0
14752 setsockopt(5, SOL_SOCKET,
Bogdan,
Thank you very much, the problem was because I forgot to add the module
dialplan in the modparam db_url:
Before:
modparam(domain|alias_db|auth_db|usrloc|drouting,db_url,
mysql://user:u...@localhost/db)
After:
modparam(domain|alias_db|auth_db|usrloc|drouting*|dialplan*,db_url,
Hi Antonio,
It should not work on any OSmaybe your mysql setups are different on
the OSs you tested. NOTE that db_url has a default value which may work
if the DB was created with default access users.
Regards,
Bogdan
Antonio Anderson M. de Souza wrote:
Bogdan,
Thank you very much, the
Hi Daniel,
Daniel Goepp wrote:
I'm going to look the complete fool here for my lack of understanding
how OpenSIPS would handle this via it's route branching, but I am just
banging my head on this one. Here is the problem:
For simplification I'm using the first octet of the IP to identify
Hi YinLin,
Well, for opensips, the cps (calls per second) is the most relevant
param. And the value is higly depending on the complexity of the
configuration/routing script.
You can do a simple SBC for NAT, but also you can do (as have recently
implemented) and SBC with TLS, NAT traversal,
Hi Jeff,
should be ok this way.
Regards,
Bogdan
Jeff Pyle wrote:
Is there any danger in adapting the default config from:
if (has_totag()) {
...
}
to:
if (has_totag() !is_method(REGISTER)) {
...
}
- Jeff
On 12/31/09 7:46 AM, Olle E. Johansson
Hi Ha,
the alternate port is whatever port you want to use as secondary port by
the STUN server. 3479 is the default alternate port in STUN, so it is ok.
Regards,
Bogdan
ha do wrote:
Hi admin
http://www.opensips.org/html/docs/modules/devel/stun.html#id227269
1.3.3. |alternate_ip|
Hi Jennifer,
A Happy New Year to you too!
for CMD1 and CMD2, try from command line
(CMD1 CMD2)
or
(CMD1 CMD2)
Regards,
Bogdan
Jennifer-4 wrote:
Hi everybody and Happy New Year!!
I´m working with Opensips 1.4.
I need to send two commands simultaneously from Opensips,
Hi Ahmed,
Asterisk is confused (by sending the request pending reply to second
CALL - csesq 2) because it never received an ACK for the reply it sent
for the first call (cseq 1). What is strange is that I do not see any
ACK neither from OpenSIPS to Asterisk, nor from UAC to OpenSIPSCould
Hey Bogdan,
It looks like that fixed it. Thanks so much!
Bill
Bogdan-Andrei Iancu wrote:
Hi Bill,
Could you please try the attached patch? It seams that there was an
issue with the the probing values in the code. Let me know if the patch
does solves your problem and I will upload it
Super! I uploaded the fix on SVN.
Thanks and regards,
Bogdan
Bill W wrote:
Hey Bogdan,
It looks like that fixed it. Thanks so much!
Bill
Bogdan-Andrei Iancu wrote:
Hi Bill,
Could you please try the attached patch? It seams that there was an
issue with the the probing values in
Interesting. I think you may have mentioned this before Dan but I didn't catch
it for some reason. That makes installing media proxy on CentOS / RHEL 5.x
easier. I've found that creating an RPM to install python 2.5 (along with the
2.4 rather than upgrading it) and then using the virtualenv
Please see:
http://www.apps.ietf.org/rfc/rfc3261.html#sec-21.5
On Jan 4, 2010, at 9:53 PM, Aditya Kumar wrote:
Hi all,
I am inter working with a 3rd party SIP UA and I see they are sending 503/500.
Can any one tell me what are the cases at which a SIP UAS will sent 503 or 500
Thanks Jeff for the link.
I did see 3261.
Question is : RFC will tel high level about response codes.
I am trying to understand in particular when a End User (UAS) will sent.
I am clear from a proxy perspective.
Was looking some one can share your experiences when a UAS will send 5xx
Aditya,
That all depends on the switch. Asterisk sends a 503 if you look at it the
wrong way, that is, under many conditions. In my experience most carriers will
send a 503 when their upstream paths are full (or your concurrent calls to them
is full) and they expect you to route advance to
Hi,
I want to implement ACL on OpenSIPs to accept the call on behalf of source
URI + IP address. Can anyone tell me which modules and functions are
required for it?
Also kindly share some example template with it.
--
Regards,
Ahmed Munir
___
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