Hello,
In the current implementation, the display name is current not saved,
and obviously, not restored.
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/17/2012 11:26 AM, Steven Lam, KeenSystems B.V. wrote:
Hi,
Thank you for your answer!
You are right, this
Hi list,
I've just install from Debian repository version 1.8.0-1-87541
(trunk) and when I try to start it fails with the next output :
Not starting opensips: invalid configuration file!
-e
Feb 29 12:38:01 [16974] WARNING:core:warn: warning in config file,
line 50, column 13-16: tls support not
Hi Steven,
The reason for not storing/restoring the display name is simple: this
information has no value in regards to the SIP dialog (it does not
affect or interfere with the dialog state, as the FROM URI does) and it
is used only in the initial invites - to be displayed when receiving a
ca
Hi Seattle,
The warning lets you know that the debs were not compiled with TLS
support - and from script, I guess, you are trying to do some TLS stuff.
The rest of the errors (about not finding certain functions in script)
are related to some recent change on trunk (keep in mind, 1.8.0 is tru
Hi there.
I've just installed OpenSIPs + OpenSIP-CP, but everytime I try to "Apply
Changes to Server" or "Refresh Dialog List", I get the following messages.
Feb 28 16:27:46 vm-opensips /usr/sbin/opensips[5079]:
ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available
Feb 28 16:27:51 vm-
Hello,
In your OpenSIPS script, do you have the dialog, dialplan, dispatcher,
domain or drouting modules loaded ?
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/29/2012 02:21 PM, Alexandre Keller wrote:
Hi there.
I've just installed OpenSIPs + OpenSIP-CP, bu
Hello, Razvan
I have try to use psevdo variable in opensips (1.7.2) script like this:
t_replicate($du,"0x4");
But i catch a bad arguments error:
CRITICAL:core:yyerror: parse error in config file, line 197, column 26-27:
bad arguments for command
Is this functional
"t_replicate can also rece
Hi.
Thanks Vlad. My mistake. I forgot to uncomment the loadmodules lines.
Everything is Ok now.
--
Atenciosamente,
ALEXANDRE KELLER
http://twitter.com/alexandrekeller
http://www.facebook.com/alexandre.keller.BR
"Dinheiro é a consequência de um trabalho bem feito e não o motivo para se
fazer
hi,
With that it works fine
Regards
On 29 February 2012 12:58, Bogdan-Andrei Iancu wrote:
> Hi Seattle,
>
> The warning lets you know that the debs were not compiled with TLS support -
> and from script, I guess, you are trying to do some TLS stuff.
>
> The rest of the errors (about not finding
Hi, Alex!
Yes, the t_replicate function hasn't changed. Your problem is that you
are not passing properly the first parameter of the t_replicate()
function. It should be:
t_replicate("$du", "0x4");
Regards,
--
Răzvan Crainea
OpenSIPS Developer
http://www.opensips-solutions.com
On 02/29/20
Yes!
Răzvan, Thank you.
Alex
uiptel.com
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/Using-dispatcher-and-t-replicate-tp5592900p7329418.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
_
I'm sorry for bothering you again.
I've trying to add some rule to my DialPlan through OpenSIPs-CP, but, it keeps
me showing the following error:
Failed to issue query, error message : MDB2 Error: syntax error
Any ideas?
Thanks again in advance.
--
Atenciosamente,
ALEXANDRE KELLER
http://t
Hi,
This is a strange error, considering the fact that I am guessing you
already have DB connectivity for CP.
Can you paste the rule that you are trying to add ? Maybe we can
replicate the error.
Regards,
Alex
On 02/29/2012 04:57 PM, Alexandre Keller wrote:
I'm sorry for bothering you agai
Hi,
I am trying to get familiar with Opensips (1.7.1) so I set up two boxes with
SIPp, OpenSIPS, and Asterisk:
box 1 (ip: 192.168.1.57) running SIPp and Opensips
box 2 (ip: 192.168.1.121) running Asterisk
I am trying to get OpenSIPS to run as a proxy between SIPp and Asterisk, but
this is the pr
Hi there,
My guess is the fault is in the sipp script - the ACK is not properly
generated : instead of using the route set information from the 200 OK
(the contact and RR URIs), it is simply sent with the same RURI as the
INVITE - this is of course bogus.
If you want, I can send you an worki
Hi David,
Here it is.
Regards,
Bogdan
On 02/29/2012 08:23 PM, David Yat Sin wrote:
Hi Bogdan,
Thanks for looking into this. Could you send me your SIPP UAC file.
Regards,
David
On 12-02-29 1:20 PM, "Bogdan-Andrei Iancu" wrote:
Hi there,
My guess is the fault is in the sipp script - the
I might be doing something wrong. I'm kind of new on OpenSIPs, I've only worked
with Asterisk for the past 6 years.
Here is my Rule, it's a testing rule.
Dialplan ID: 1
Rule Priority: 1
Matching Operator: REGEX
Matching Regular Expression: 123*
Matching String Length: *
Substitution Regular Expr
Greetings!
The ClueCon team is gearing up for this summer's event, which will be held
August 7-9, 2012 at the Wyndham in downtown Chicago. We want to hear your
proposals for presentations to be given at ClueCon '12. Here's what we're
looking for:
- Discussions on technology or software th
When I setup a call then cancel the call I am getting a 487 from my gateway
that is relayed to the client.
I don't wish to show the 487 to the client. Is it possible to drop the reply
for the 487?
I am hoping someone could let me know if this is possible.
I am running opensips 1.7 Here is
Hi Sammy,
Sure I will paste the stats here but will you please tell me some important
steps to be performed on software side. Here I am talking about the OS, kernel
upgrade & advanced system administration level stuff that can reduce the packet
loss immediately. Through the final conclusions I
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