Hi Bogdan,
Could you give a link to MI2FIFO proxy from ag projects?
Can't find it :(
> -Original Message-
> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> Sent: Friday, May 06, 2011 9:19 PM
> To: users@lists.opensips.org; Anton Zagorskiy
> Subject:
mi_xmlrpc log
(i've setted modparam to /var/log/abyss.log)
Any suggestions?
--
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
e-mail: a.zagors...@oyster-telecom.ru
tel:+7 812 601-0610
fax:+7 812 601-0
sing 'real' errors.
Also, it will be very helpful, if B2B says that it received request for
already finished session.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Mes
Hi Duane.
In most cases it means that it can't connect to DB.
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of
duane.lar...@gmail.com
Sent: Tuesday, April 12, 2011 3:44 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] OpenSIPS-CP - Call
Hi Anca,
Thanks.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensip
> Hi.
>
> I have dedicated OpenSIPS only for B2B Top Hiding.
> What kind of SIP check should I do in a request route? Same as I'm doing
in a
> 'typical' OpenSIPS config?
>
> I'm asking because of cfg is quite (very quite!) simple but periodically
I'm
> getting in the log
> ERROR:b2b_entities:b2b_
entities:b2b_tm_cback: No dialog found reply 200 for method BYE
Also, I'm getting a situation when a request has TO-tag but isn't
loose_route (this occurs when B2B didn't found dialog - strange too).. why
OpenSIPS doesn't reply Call leg/Transaction Doesn't Exist?
WBR, An
Hi Bogdan,
thanks for answer and link.
I asked because of while re-reading Flavio Goncalves's book (OpenSIPS 1.6)
I've found that he handle failure route on RE-INVITE and thus his scripts
aren't correct sometimes.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.:
Hi.
Does it needed to handle RE-INVITE by failure_route?
We can't use drouting module and we shouldn't redirect a RE-INVITE to media
services.
What we can do with RE-INVITE in the failure_route?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax:
Bogdan,
I've never meet such situation, so I don't know what to do...
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: Bogdan-Andrei I
r. How can I do this if I don't know
request user?
There is no any data with given Call-ID in log between BYE and failure
request. Failure route was totally unexpected...
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-t
te_avp: 0 avps were removed
DBG:core:comp_scriptvar: str 20 : 0
DBG:tm:t_check_status: checked status is <408>
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
__
Bogdan,
thank you again!
> -Original Message-
> From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> Sent: Tuesday, April 05, 2011 11:31 PM
> To: Anton Zagorskiy
> Cc: 'OpenSIPS users mailling list'
> Subject: Re: [OpenSIPS-Users] uac_replace_from an
Hi Bogdan,
> > Where is a problem?
> The problem is that whatever you do in the main request route does apply
to
> all the branches (existing or future) of that request. So you
> uac_replace_from (from 1) does apply to all branches, including to the
> branch you create in failure route).
>
> The
Hi Bogdan,
Thank you, its working.
What about my first question - mixing two openSIPS on a same DB where second
openSIPS is just for b2b top hiding?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster
ot;)
2) FAILURE_ROUTE: failure_route[1] -> route[failure] -> route[invite]
(uac_replace_from) -> route[to_b2b] (uac_replace_to) -> t_relay with
t_on_failure("1")
And here I'm getting bad headers:
Record-Route has two 'vsf' tag and From is "num1" "num2"
> all (see my comment above). But anyhow, adding a IP to the "domain"
> table doesn't produce a loop at all, so it must be an error in your
> conf.
I had the same situation - openSIPS goes to ACK looping to itself when OK
received when there is an IP in the domain table.
When I've changed IP to
]: DBG:db_mysql:db_mysql_do_prepared_query: doing BIND_PARAM in...
[46968]: CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error
(1048): Column 'to_tag' cannot be null
[46968]: ERROR:dialog:dialog_update_db: could not add another dialog to db
Is it normal?
WBR, Anton Zagorskiy
VoIP Developer, Oyst
wn opensips:opensips /var/run/opensips
Never had that issue.
cd /var/run
ls -la
drwxr-xr-x 2 opensipsopensips4096 Feb 24 08:23 opensips
On Thu, Feb 24, 2011 at 3:07 AM, Anton Zagorskiy
wrote:
Hi.
I've made a user opensips with group opensips and with shell nologi
start with:
ERROR:core:daemonize: unable to create pid file
/var/run/opensips/opensips.pid: Permission denied
A command "sudo -u opensips touch /var/run/opensips/opensips1.pid" works
well.
What I'm doing wrong?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax:
prefix("12345#");
or
$rU = "12345#" + $rU;
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Brian Artigas
Sent: Friday, February 18, 2011 5:58 PM
To: users@lists.opensips.org
Subject: [OpenSIPS-Users] Add characters to uri
Hello,
I have a provi
ilable.
#4 0x00080085c17c in gethostbyname () from /lib/libc.so.7 No
symbol table info available.
#5 0x0047616c in fix_socket_list (list=0x65da08) at
resolve.h:349
si = (struct socket_info *) 0x69e770
l = Variable "l" is not available.
WBR, Anton Zago
SIPS replies "401: Forbidden" but continues the execution of the
> script.
>
> Is this normal?
>
>
>
>
>
>
> WBR, Anton Zagorskiy
> VoIP Developer, Oyster Telecom
> Phone.: +7 812 601-0666
> Fax: +7 812 601-0593
> a.zagors...@oyster-telecom.
ot;401: Forbidden" but continues the execution of the script.
Is this normal?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
___
Users m
Hi Toyima.
I see 200 OK reply. What is wrong? =)
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Toyima Dias
Sent: Thursday, February 03, 2011 11:32 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] authentication is not working
Hello my
Hi.
Is it possible to log not to a 1 file via xlog?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
___
Users mailing list
Users
Hi.
Kamailio hasn't B2B module.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Toyima
heck_routing: no rules => allow any routing" and
nothing more from permissions module.
In each file there is at least one rule.
What I'm doing wrong?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.z
Hi Dave.
> So just to be sure I'm clear on this. $Ts rounds down ( truncates ) the
current second. So using $avp(s:start_time) = $Ts.$Tsm; would give something
like "12343253.543233" and always be accurate?
You should understand that $Ts.$Tsm is two independent calls to the system
time function.
Hi.
Please help me with advice.
I want to do a call limitation based on prefixes for each user in each
domain. In other words, I need to deny or allow call depend on username and
domain and value on a field in a sql table.
Which module should I use?
Thanks!
WBR, Anton Zagorskiy
VoIP
Thank you, Olivier!
2Bogdan: this update fix my problem about B2B and '404' reply.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
>
here"
On Wed, Jan 19, 2011 at 4:21 AM, Anton Zagorskiy
wrote:
Bogdan, you are right!
I sent 404 from script because of BYE, that was sent from B2B to openSIPS is
has_totag() but not loose_route()
Just a quick comment..
"404 Not Here" is generated by the top of the script and not by
nderstood from you that you are using 2 opensips instances - one as
> proxy, another one as b2bua - is this correct ?
>
> The script you posted here , which one is ?
>
> maybe having a sip capture of the call will help a bit (taken from both
> opensips)
>
> Regards,
>
Hi, Andrew.
Maybe I'm wrong, but using only acc module it is impossible.
I'm using dialog module and manually store start and end time of a duration.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
route(1);
exit;
};
if (is_method("REGISTER"))
{
if (!save("location"))
sl_reply_error();
exit;
};
}
route[1]
{
if (!t_relay())
{
sl_reply_error();
exit;
};
}
sent from script. I suspect your
> sequential requests are not properly handled and it hits the
> lookup(location) (which is for initial requests).
>
> Regards,
> Bogdan
>
> Anton Zagorskiy wrote:
> > Hi Bogdan.
> >
> > I fixed the B2B loop, but openSIPS stil
t;
>
> And, openSIPS replies to B2B '404'
>
>
> I see 2 potential moments:
> 1) I'm changing 'From' field on stage 2.
> 2) On stage 3 a field 'Contact' is "Contact:
> " althought I defined
> "modparam("b2b_lo
ge 2.
2) On stage 3 a field 'Contact' is "Contact:
" althought I defined
"modparam("b2b_logic", "server_address", "sip:sa@192.168.0.10")"
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-066
url", "mysql://login:pass@host/db")
modparam("acc", "cdr_flag", 2)
...
Later in the script
setflag(2);
setflag(10);
$dlg_val(..) I'm setting up during the script.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7
As I told, new B2B INVITE hasn't original Via headers. Is it normal?
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
> Sent: Monday, January 17, 2011 7:53 PM
> To: 'Ope
Also, B2B sends INVITE where there are 2 Via headers which point to openSIPS
IP address and there is 1 Record-route which points to openSIPS IP too.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster
19:01:25 softswitch [softswitch][9330]: DBG:core:parse_msg: reason:
Jan 17 19:01:25 softswitch [softswitch][9330]: DBG:core:parse_headers:
flags=2
While debugging at line marked *** I see that warning_len is 0
Any suggestions?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0
nuary 17, 2011 3:26 PM
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] milliseconds and random values
>
>
> On 01/17/2011 12:19 PM, Anton Zagorskiy wrote:
> > Inspection source code I've found a $Tsm pseudo variable that returns
> > milliseconds of a
to:users-
> boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
> Sent: Friday, January 14, 2011 6:59 PM
> To: 'OpenSIPS users mailling list'
> Subject: [OpenSIPS-Users] milliseconds and random values
>
> Hi.
>
> I want to use milliseconds of current time and ran
Hi.
I want to use milliseconds of current time and random values in the config
file.
$time doesn't support milliseconds and I didn't find any function that
generates random value.
Is it any way to do that?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
F
going lines and accounting (billing)
doesn't work properly (there is just one leg between A and C)
I haven't any idea how to solve this situation.
B2B could help me by means of sending RE-INVITE to A and C when B sends
REFER, instead of transmitting REFER to A.
WBR, Anton Zagorskiy
the site? There are 1.6.3
and trunk/devel in Cookbooks and 1.6.x and 1.7.x in the tutorials. What is
1.7.x?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
scenario from b2b tutorail? In this case
method_value is METHOD_REFER.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.o
order for you to use b2b the SIP message cannot have
a ToTag.
You have to use B2B like so
if(is_method("INVITE") && !(src_ip == "OPENSIPS_IP") && !has_totag() ) {
{
location();
b2b_init_request("refer");
exit;
}
On Tue, Dec 28, 2010 at
b2b_init_request("refer");
xlog("*** route: REFER request, B2B(refer) called,
exit");
exit;
};
...
};
};
};
What I'm doing wrong?
WBR, Anton Z
e was
calling. But failure route wasn't calling in the original transaction. Why?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
___
Users ma
Also, is it possible to force openSIPS to save _all_ internal data to a DB?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: user
drouting module:
The migration script alters column 'groupid' in the table 'dr_rules' to the
type int, but drouting module checks it as DB_STRING (see dr_load.c line
536)
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-059
Hi.
I see you removed diaplan from default modules. Could you explain why?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: user
eiver udp:92.255.16.115:5060
Process:: ID=6 PID=4797 Type=time_keeper
Process:: ID=7 PID=4798 Type=timer
Process:: ID=8 PID=4799 Type=MI FIFO
The evil process was 4795
I've tried to use gdb awatch but it didn't help.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.
Hi.
Please see dumps in the attach
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
I've made a simple config, and openSIPS still crashes.
Please see my config here http://pastebin.com/mfabxGJJ
And please see a full log file http://pastebin.com/0VXQ6vwT
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@o
Hi!
What about investigation my crash?
Should I send any additional information?
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Anton Zagorskiy
> Sent: Tuesday, December 14, 2010 5:01 PM
> To:
reply("2");
t_on_failure("1");
};
if (!t_relay())
{
sl_reply_error();
};
xlog("*** --- route[main_route] has finished");
}
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@
: done
[8032]: DBG:core:parse_msg: SIP Request:
[8032]: DBG:core:parse_msg: method: (# The same INVITE is
coming!)
What I'm doing wrong?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.
Make logging to a file and set debug=9.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
Hello.
It seems that when I call do_route() the route[1] was automatically called..
As I understand I should manually call relay functions after do_routing().
Any ideas why that is happening? :)
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601
The problem was that I defined variables as integer and assigned a string
value to them later.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
From: users-boun...@lists.opensips.org
ITICAL:core:comp_scriptvar: cannot get left var value
[30629]: WARNING:core:do_action: error in expression (l=391)
What a difference between line 388 and 391 so 391 raises the error?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7
I'm using drouting 1.6.3
Module's documentation say nothing about 'attrs' in the dr_rules table. (See
http://www.opensips.org/html/docs/modules/1.6.x/drouting.html#id250026 )
Is it possible to re-create drouting's tables?
WBR, Anton Zagorskiy
VoIP Developer, Oyster T
dr_reload_data: failed to load routing info
[30268]: CRITICAL:drouting:dr_child_init: failed to load routing data
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
___
Hi Bogdan.
What about when $DLG_status is 1 ? Was I right in my previous letter?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: user
2 in doc. But
> as I understand status 2 exists during time after create dialog and
> until final reply.
> Then status can be 3,4 (if final reply received) or 5(if there is no
> final reply).
>
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users
So, can you please update the documentations? I should know that status 2
isn't temporarily status from developing branch and it wouldn't disappear
without any comments in a further version.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 81
When INVITE is sent and OK isn't received $DLG_status equals 2. But in the
documentation $DLG_status can be NULL, 3, 4, 5.
When does $DLG_status is 2? Can it has other values?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@o
Hello.
I'm using a dialog module and store values in dialogs. Is it possible to
fetch values from dialog while ACC logging?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telec
this:
route[check_parameters]
{
xlog("*** +++ route[check_parameters]");
}
timer_route[check_parameters, 3]
{
xlog("*** +++ timer_route[check_parameters, 3]");
}
But this doesn't work - route[check_parameters] didn't called. Why?
>
> Regards,
>
Hello.
Do cache_fetch and cache_store functions have any locking mechanism to
prevent race condition?
$shv variables?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
Tue, Nov 30, 2010 at 11:19 PM, mayamatakeshi
wrote:
2010/11/30 Anton Zagorskiy
Why BYE isn't a part of the transaction?
>From RFC3261:
Specifically, a SIP transaction consists of a single request and any
responses to
that request, which include zero or more provisional respo
Why BYE isn't a part of the transaction?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Beha
ethod("BYE"))
{
xlog("*** BYE $avp(i:2)");
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
___
Users mailing list
> how do you load these params at startup and where do you store them
> (what kind of variables)?
>
> Regards,
> Bogdan
>
> Anton Zagorskiy wrote:
> > Hello.
> >
> > I have some parameters that are stored in a DB. I'm loading it once
> time
> >
Nobody uses limitation on amount of established sessions? I don't believe!
=)
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From:
transaction is in the terminating stage?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
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Thanks, this is work.
Can you explain why rewritehost() doesn't accept AVP? Where AVP doesn't work
too?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Messag
mn 38-39:
bad argument, string expected
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
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Hello.
I have some parameters that are stored in a DB. I'm loading it once time
when opensips starts.
After when they are changed in the DB I want to re-read it. How can I do
this without restarting opensips?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-066
So I need to change db_url parameter through all modules.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.org [m
Thanks.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On
error too.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
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Hello.
Can I change module's parameter in the route block? I want to change db_url
parameter.
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telec
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