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 Iancu [mailto:bog..
Anton,
What method is the one you end up with no username in failure route ?
Regards,
Bogdan
On 04/06/2011 07:05 PM, Anton Zagorskiy wrote:
Hi Bogdan,
Such situation (when in the failure route $rU is NULL) isn't good - I've
checked log using Call-ID, and:
1. I already received BYE on that ses
Hi Bogdan,
Such situation (when in the failure route $rU is NULL) isn't good - I've
checked log using Call-ID, and:
1. I already received BYE on that session (and stored CDR)
2. I'm using call forward on request timeout and use $rU to determine
forward sip account of dialed user. How can I do this
Hi Anton,
In sequential requests you typically have in RURI the contact URI from
the original INVITE + 200 OK. And username in contact URIs in most of
the case irrelevant - important is the IP, port and proto.
Regards,
Bogdan
On 04/06/2011 06:23 PM, Anton Zagorskiy wrote:
Hi.
In which case
Hi, thanks for the reply. The issue was that my comparison was being treated as a string instead of an integer so 2 > 15. Once I converted to integers this looked to be resolved:$avp(s:sim_calls) = $(avp(s:sim_calls){s.int});$avp(s:cnt) = $(avp(s:cnt){s.int}); if($avp(s:cnt) >= $avp(s:sim_calls)) {
Hi Chris,
The problem is not in triggering the event in script when a dialog
timeout occurs - the problem is that all events in opensips are SIP
message driven :)so to be able to run a script route, you need a sip
message.
Regards,
Bogdan
On 03/15/2011 01:10 PM, Chris Martineau wrote:
Hi.
In which cases in the failure route will be passed a request with $rU ==
null?
In a log I see just:
DBG:tm:utimer_routine: timer routine:4,tl=0x803ad3218 next=0x0,
timeout=25630
DBG:tm:timer_routine: timer routine:2,tl=0x803ae43c0 next=0x0, timeout=260
DBG:tm:wait_handler: removing 0x803a
Sorry for the confusion. Orginally when I was testing my opensips config
with the dialoginfo_set() function used in the script I was running into
opensips memory (pk and sh) running out. This is most likely due to the
fact that a ton of records were building up in the PUA table because
open
Hi Logan,
what are the printed xlogs from your script ? interesting is to see the
values for $avp(s:cnt) and $avp(s:sim_calls) .
Regards,
Bogdan
On 03/15/2011 04:15 PM, logan wrote:
I have a uuid in the usr_preferences table set to attribute =
sim_calls with value = 15 calls allowed but am
Hi Shawn,
If you say that the problem is with the contact in 200 OK, maybe you
should address it there : when handling replies in onreply_route, if the
contact IP is private ( use nat_uac_test() -
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id294025
, test "1") replace it w
Hi John,
First of all I'm trying to see if \n\n is allowed in the username value
(which is a quoted-string). According to the SIP grammar :
A string of text is parsed as a single word if it is quoted using
double-quote marks. In quoted strings, quotation marks (") and
backslashes (\)
Hi Bogdan,
The "inversion" is only in the triggered function.
The arguments are passed correctly from left to right to the fixup function.
The triggered function is:
test_function(struct sip_msg *msg, char *arg1, char *arg2, char* arg3, char*
arg4, char* arg5, char* arg6)
{
str arg1_s = {N
Hi Denis,
On 03/21/2011 01:55 PM, Denis Putyato wrote:
Hello
There is such scheme of call
One gateway – 1.1.1.1
Opensips – 2.2.2.2
Another gateway – 3.3.3.3
Calls from 1.1.1.1 to 3.3.3.3 through 2.2.2.2
I use CDR_flag for accounting
As you can see in testlog file, 1.1.1.1 trying cancel i
Hi Denis,
I agree with you on the 4th xlog - it should be printed only if the BYE
matched the dialog - of course, if you identified the xlog sequence
correctly :)
If the BYE matched the dialog, then the dialog should be terminated and
no timeout.
What I can think of is that the BYE is not
Duane,
trying to follow what you are saying :) - the mem leak you observe is
for system memory (when you look with top) or is opensips memory
(private or shared) ?
Regards,
Bogdan
On 04/06/2011 06:18 AM, Duane Larson wrote:
OK after more testing it looks like my current test config with
dia
Hi Bogdan,
This "inversion" is visible in the fixup function or is the triggered
function ?
Regards,
Bogdan
On 04/06/2011 02:44 PM, rad bogdan wrote:
Hi everyone,
I implemented an OpenSIPS custom module that communicates with an
external application. From the routing script I call a functi
Hi everyone,
I implemented an OpenSIPS custom module that communicates with an external
application. From the routing script I call a function with 6 parameters:
test_function('$fU', '$tU', '$ci', '$avp(s:name1)', '$avp(s:name2)',
'$var(name3)')
The problem is that when the corresponding funct
Hi Bogdan,
Thank you! I'm now putting now ANI in into match_exp field and the
prefix into attrs - works like a charm.
On 05.04.2011 22:36, Bogdan-Andrei Iancu wrote:
But why using the ANI as dpid and not using it as regexp part in the
rule ? See:
http://www.opensips.org/html/docs/modules/1.6.x
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 and two openSIPS 1.6.4
>
> Hi Anton
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