Congratulations Bogdan, developers, testers and implementers that made this
possible!!
--
Zahid
On Mon, Oct 9, 2017 at 7:05 AM, Bogdan-Andrei Iancu
wrote:
>
> We are all proud to announce that the OpenSIPS project is a winner of the
> Google Open Source Peer Bonus - this is an official recogn
i-sip-timer.html
Regards,
--
Zahid
On Mon, Jan 11, 2016 at 6:03 AM, Serge S. Yuriev wrote:
> Zahid,
>
> Would you mind to share us *secret* option for cisco please?
>
> On 06/01/16 18:56, Zahid Mehmood wrote:
>
>> Hi,
>> INFO is generated in early stage.
>>
&
erface to wait for the FACILITY message before sending the
>> initial INVITE. When the INVITE does leave the gateway towards the proxy,
>> it has full caller name information. Perhaps something like this is
>> available on the Cisco. I hope so, because if not, you're goi
> it has full caller name information. Perhaps something like this is
> available on the Cisco. I hope so, because if not, you're going to have a
> difficult time integrating the INFO message.
>
>
> - Jeff
>
>
> On Thu, Dec 17, 2015 at 2:53 PM, Zahid Mehmood wrote:
&
Hi,
I am having trouble figuring out how to process the calling-name coming
from the PRI. In my setup, PRI is connected to a Cisco media gateway which
sends traffic to the proxy servers. Calling name is not coming in the
ISDN setup message. It is actually provided in a separate facility messa
Hi Bogdan,
This is cool. Thank you for all the hard work. I look forward to
using this.
Best regards,
--
Zahid
On Wed, Jul 31, 2013 at 1:32 PM, Bogdan-Andrei Iancu wrote:
> Hello,
>
> The presence_callinfo module was extended to provides OpenSIPS full
> support for shared call appea
Bogdan,
I'm wondering if this is something on the list of things to do.
The most recent version that I've tested is 1.7.2.
Thanks.
--
Zahid
On 09/01/2010 09:05 AM, Bogdan-Andrei Iancu wrote:
yes and I guess the issue is that when using TM as UAC, the DNS-based
failover is not done, lik
Did you restart syslog after the change?
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ali Pey
Sent: Wednesday, June 06, 2012 3:05 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Call per second - OpenSIPS performance
Brett, Duane
Chris,
Are you currently using xlog to write messages to log file? If yes, make
sure that you configure syslog to use asynchronous logging for the log facility
used by opensips.
--
Zahid
On Feb 21, 2011, at 9:06 AM, chris wrote:
Hi,
I assume you mean increase the log level above 3 wh
Thanks Anca and Bogdan. Yes indeed that is the desired functionality. In its
current state, BLA breaks for us if one proxy is unavailable and opensips
happen to pick that one to send presence related traffic .
--
Zahid
On Sep 1, 2010, at 9:05 AM, Bogdan-Andrei Iancu wrote:
> Anca Vamanu wrot
Anca,
It may not be happening in presence/pua code but I think OpenSIPS does
perform the lookup at some point to get the IP to build the actual packet to
send out.
--
Zahid
On Aug 30, 2010, at 7:55 AM, Anca Vamanu wrote:
> On 08/27/2010 07:33 PM, Zahid Mehmood wrote:
&g
HI Sujeev,
try this in place of your append_hf statement.
if(is_present_hf("Remote-Party-ID"))
{
remove_hf("Remote-Party-ID");
}
if(is_present_hf("Privacy"))
{
remove_
Hi,
With pua_bla's default_domain set to point to use a naptr record (srv
records resolve to two hosts), opensips picks one hostname/IP and only attempts
to send sip messages to it. If that host is unavailable, OpenSIPS does not
attempt to contact the second host returned by the SRV recor
See if this can open OpenSIPS VM (or the disk)
http://www.virtualbox.org/wiki/Downloads
--
Zahid
On Aug 4, 2010, at 12:07 AM, David J. wrote:
> Turns out vmware is not free for Mac. Guess I have 30 days to play
> before trial runs out.
>
> Maybe we can make a livecd instead not vmware bas
erfect Zahid. I was just starting to build a scenario for
> testing BLA. Do you have one already built? If so can you share it with the
> list?
>
> Thanks,Steve
>
> From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.or
Take a look at the "Measuring REGISTER performance using SIPP" found at
http://www.opensips.org/Resources/PerformanceTests.
Hope this helps.
--
Zahid
On Apr 21, 2010, at 5:49 AM, Alireza6918 wrote:
>
> Hi
> I want to test opensips by sipp.
> can anyone please help me?
> --
> View this messa
Philipp,
The accounting (acc) module can be used to log extra values. You can
configure the module to save the billing party in acc records and generate cdr
using that information.
(http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id271945 )
--
Zahid
On Apr 5, 2010, at 7:23 AM
Steve,
alter db table "pua" to add a new column "to_uri". That should do it. I
don't think it is documented yet.
--
Zahid
On Mar 2, 2010, at 1:19 PM, Steven C. Blair wrote:
After upgrading to the svn head release I get the following error when starting
opensips with my previously workin
Anca,
I'm sending you packet captures directly.
This is what I tried.
- 3 phones configured for BLA (using same aor and user credentials)
- phone A and phone B are online.
- 1st incoming call answered on phone A and placed on hold
- 2nd incoming call answered on phone B and placed
"Reason: SIP ; cause=200"
header when CANCEL in forking scenarios
El 09/02/10 16:01, Zahid Mehmood escribió:
> Hi,
> Wondering what was the end result of this.
>
> Was this put in as a feature request? or is it implemented already?
>
> Thanks.
>
It was impl
Hi,
Wondering what was the end result of this.
Was this put in as a feature request? or is it implemented already?
Thanks.
--
Zahid
On Jan 21, 2008, at 11:26 AM, Bogdan-Andrei Iancu wrote:
> Hi Iñaki,
>
> Right now this RFC is not supported, but it is very doable - you just
> ne
John,
See if this helps.
http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id271094
1.5.5. detect_direction (integer)
Controls the direction detection for sequential requests. If enabled
(non zero value), for sequential requests with upstream direction
(from callee to caller), th
Noel,
Make sure that you syslog is configured to do async write to the
log file.
--
Zahid
On Sep 21, 2009, at 7:24 PM, Noel R. Morais wrote:
> Hi Guys,
>
> I was making some stress tests and I realized that when I let the
> syslog enable, the opensips becomes really slow.
>
> In my tes
Hi,
I need some guidence in handling a "300 Multiple Choices" from my
media gateway (Cisco).
The scenario is as follows:
1. phone1 --> media gateway --> opensips proxy --> phone2 (answered)
phone2 needs to transfer the call to phone3
2. phone2 --> opensips --> media gateway --> phone3 (a
Are your media gateways (or end points) configured for T38? I had to
configure the Cisco media gateway to support T38.
Have you looked at packet capture to see where it is failing?
I was able to see a negative reply from the gateway when the fax server
tried to re-invite to negotiate T38.
-
I use the following on RHEL4, using logrotate, and it works fine.
--
/var/log/openser.log {
sharedscripts
postrotate
/bin/kill -HUP `cat /var/run/syslogd.pid 2>/dev/null` 2>/dev/null ||
true
endscript
}
--
Zahid
-Original Message-
From: users-boun...@lists.opensip
Forgot to include the list before
On Feb 26, 2009, at 2:42 PM, Yehavi Bourvine wrote:
2009/2/26 Zahid Mehmood
i tested directed pickup and it worked fine in pure sip
environment.. the only issues i had were with the cisco media
gateway not working properly with REFER etc
If you are still using the Polycom phones, you can play with the
"group-call-pickup", "directed-call-pickup" features. Enabling these
two will make two soft keys visible when you make the phone go off-
hook. Essentially the phone will use subscribe/notify to find out
the state and then use
Hi Gonzalo,
Is lookup("location") being called before the uri matching or after?
Looking at the code snippet, If local callers are dialing a pattern that
matches one of the "if (uri=" Then it is sent to route(4) and
lookup part of the code is not touched.
You may want to add "x
Hi Yehavi,
When using BLA, polycom phones append p-preferred-id hdr. Use opensips
to replace that with a p-asserted-id hdr.
To show who answered the call, set t_on_reply when processing the
initial invite, and use on_reply_route to insert the header.
HTH.
--
Zahid
_
Yehavi you can run a packet capture on your presence server and then
start the phone. This is the sequence of sip message you should see
if your phone and proxy/presence configuration is correct:
1. REGISTER coming from the phone to the presence server (may be
t_replicated from the proxy
Hi,
any particular reason why scripts for testing like this can not be
shared on the wiki? I think we all can benefit from having standard
tests and some benchmarks to compare against.
If for some reason this script can not put placed on the wiki, can I
request a copy?
Thanks.
--
Zah
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