Hi Muhammad,
exactly that it did, that is the problem:)
Could be wrong opensips version (1.8.2)?
Here is my pastebin: http://pastebin.com/H8jXpkub
You can see that I did exactly what you said:)
Thanks for help!
BR,
Miha
On 11/19/2012 7:34 PM, Muhammad Shahzad wrote:
Here is the sequence o
Actually, what you really need is only a new transport for MI interface.
For a json API for MySQL, take a look here:
http://rackerhacker.com/2012/03/28/mysql-json-bridge-a-simple-json-api-for-mysql/
Regards,
Ovidiu Sas
--
VoIP Embedded, Inc.
http://www.voipembedded.com
On Mon, Nov 19, 2012 at 6
So you are looking for two new things:
- a new transport for MI interface;
- a new way to access the opensips db API (similar to pi_http module).
Regards,
Ovidiu Sas
--
VoIP Embedded, Inc.
http://www.voipembedded.com
On Mon, Nov 19, 2012 at 3:43 PM, Binan AL Halabi
wrote:
> Hi Ovidiu, Thank
Hi Ovidiu, Thank you for your response.
I see your thread here :
http://lists.opensips.org/pipermail/users/2012-October/023390.html for web
based provisioning:
I mean something like that but based on JSON and REST.
But at the same time there is need for MI functions (reconfigure on the fly).
What are you looking for: to provision or to manage opensips?
The MI interface is for management, not for provisioning.
Provisioning is a simple matter of filling up opensips tables and you
can do that externally (outside opensips).
Regards,
Ovidiu Sas
--
VoIP Embedded, Inc.
http://www.voipembed
Hello all,
Any hint on this issue, please?
Thanks in advance.
Mariana.
On Fri, Nov 16, 2012 at 11:33 AM, Mariana Arduini
wrote:
> So, I tried this:
>
> modparam("dialog", "timeout_avp", "$avp(session_expires)")
> ...
> modparam("sst", "min_se", 90)
> modparam("sst", "timeout_avp", "$avp(sessio
Hi Christian,
load and configure "auth_db" and "auth" modules. Then use these function:
proxy_authorize() and www_authorize()
see these:
http://www.opensips.org/html/docs/modules/devel/auth_db.html#id250229
And chapter 5 of book "Building Telephony Systems with OpenSIPS 1.6"
// Binan.
___
Good idea. However, for now, you can use some language binding e.g. PERL or
LUA to write up your own opensips module for this support.
Thank you.
On Mon, Nov 19, 2012 at 7:50 PM, Binan AL Halabi wrote:
> Hi all,
>
> Is there a way to provision OpenSIPS using JSON format instead of XML and
> RES
Hi.
Using 2 soft-SipPhones I specify the IP-address of my OpenSIPS proxy and
manage to register 2 users that I created using the OpenSIPS control
panel
The users can call each other and everything, logging/traces on the
proxy clearly show the traffic between the 2 users
Now the strange thing is
Hi all,
Is there a way to provision OpenSIPS using JSON format instead of XML and REST
instead of XML-RPC ?
I think there should be new module named like MI_JSONREST or something like
that.
Thanks.
// Binan___
Users mailing list
Users@lists.opens
Here is the sequence of actions, which must be done in order,
1. edit syslog.conf to define your custom log file for opensips.
2. restart syslog service.
3. restart opensips.
Thank you.
On Mon, Nov 19, 2012 at 5:28 PM, Miha wrote:
> Hi,
>
> I am using centos. I have restarted syslog service
>
Hi,
I am using centos. I have restarted syslog service
(/etc/init.d/syslog restart) which created me opensips.log.
But still it is empty.
On Mon, 19 Nov 2012 14:34:31 +0100
Muhammad Shahzad wrote:
> you need restart syslog service. Under debian you can do
> this by,
>
> service rsyslog restar
After I run mediaproxy/opensips with the new config files ,I still facing such
issue ,
What kind of trouble shooting I can do ….
Regards
Khaled
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Muhammad Shahzad
Sent: Monday, November 1
Hi all, i need to do parallel forking towards 2 different systems. When one
of them had picked up the call, opensips has to be able to knows that until
call is terminated. What i need is, if a second call comes again, opensips
has to reply to the caller the busy tone.
Is this possible?
SCENARIO:
you need restart syslog service. Under debian you can do this by,
service rsyslog restart
This will create log file as well.
Thank you.
On Mon, Nov 19, 2012 at 1:44 PM, Miha wrote:
> HI,
>
> here is my pastebin when I do opensips start:
> http://pastebin.com/Z9hxrHB2
>
>
> debug=6
> log_std
Your media proxy configuration is also wrong, just sent you config for
media proxy. Just sent you config for media proxy as well.
Hope this helps.
Thank you.
On Mon, Nov 19, 2012 at 2:24 PM, M.Khaled W Chehab wrote:
> Hi,
>
> ** **
>
> Kindly note that I can still find Calls have the same
Hi,
Kindly note that I can still find Calls have the same problem (max dialog
timeout duration and the call capture mark to be a canceled call )with a label
in dialog module while the call is not hanged by system ( confirmed but not
ACK),as for the avp queries in my script you can delete
HI,
here is my pastebin when I do opensips start:
http://pastebin.com/Z9hxrHB2
debug=6
log_stderror=yes
log_facility=LOG_LOCAL0
syslog.conf
#opensips
local0.* /var/log/opensips.log
Still no luck:(
I tried to cat massages, opensips.log but there is no opensips log.
Br,
Miha
On 11/19/2
Hi all.
It seems that stable18 in the project's Debian package repository
(apt.opensips.org) does not yet contain packages for OpenSIPS 1.8.2. Is
that correct?
One of our testing systems has been installed some time ago, when 1.8.0
was current. Since in the meantime 1.8.1 and 1.8.2 have been rele
Hello,
Is it possible to terminate/cancel a call in local_route?
If yes, how?
Thank you!
Jorge Pinho
Analyst/Developer
Network Platforms and Multimedia Solutions
Multimedia Division
___
Users mailing list
Users@lists.opensips.org
http://lists.opens
On Nov 19, 2012, at 12:26 PM, Miha wrote:
> Hi,
>
> my opensips.cfg:
> debug=6
> log_stderror=yes
> log_facility=LOG_LOCAL0
>
> syslog.conf
>
> #opensips
> local0.* -/var/log/opensips.log
>
>
> Where could be a poroblem that opensips does not log?
> Opensips.log is empty.
>
Remove that '-'
Hi,
my opensips.cfg:
debug=6
log_stderror=yes
log_facility=LOG_LOCAL0
syslog.conf
#opensips
local0.* -/var/log/opensips.log
Where could be a poroblem that opensips does not log?
Opensips.log is empty.
Thanks!
BR,
Miha
___
Users mailing list
Use
Thanks I found it
Regards
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of M.Khaled W Chehab
Sent: Monday, November 19, 2012 11:35 AM
To: 'OpenSIPS users mailling list'
Subject: Re: [OpenSIPS-Users] calls have max call duration
Mohammad,
Mohammad,
There is no file attached ,kindly can you resend it .
Regards
Khaled
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Muhammad Shahzad
Sent: Sunday, November 18, 2012 9:38 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenS
Hello Opensips community,
I'm facing an issue using opensips (v1.8.2) + rtpproxy.
opensips is used as a SIP proxy (+ NAT traversal).
another component is handling the service logic (Application Server)
The call scenario is the following:
- 2 UAC are registered using the same identity (using the s
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