Hi Bogdan,
Many thanks for your immediate help:)
Best wishes,
Chen-Che
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Hey all,
I'm trying to create a custom e164 class in cdrtool. I started out
with just uncommenting the default example one in global.inc that is
just a copy of the real E164_US class found in cdr_generic.php; it
looks like:
class E164_US_Custom extends E164 {
function E164_US($intAccessCode='0
In the meantime, I have disabled the qos module from my production
environment. Lets see if it crashes again.
--
Ahsan Hasan
On Fri, Feb 14, 2014 at 9:34 AM, Ovidiu Sas wrote:
> I just saw now that this is related to the qos module. I will take a look
> and get back to you all.
>
> -ovidiu
> O
I just saw now that this is related to the qos module. I will take a look
and get back to you all.
-ovidiu
On Feb 13, 2014 11:17 PM, "Ahsan Hasan" wrote:
> Hi,
> I have not been able to reproduce this issue in my dev environment
> (replica of production), because the only traffic on that server
Hi,
I have not been able to reproduce this issue in my dev environment (replica
of production), because the only traffic on that server is mine.
Where as on production environment, its occurrence is totally random, it
may not crash in a month, or would crash twice a day.
--
Ahsan Hasan
On Thu, F
Hi, Bogdan
Thanks for the reply, I have already add gzcompress feature to my sip server
from kamailio
that opensips use version 1.11.0, but when sip server run sometime,
server cannot accept tcp connections.
OS is CentOS 6.3 X86_64
log as below,
Feb 11 14:45:32 bogon ./opensips[30985]: INFO:cor
Dear Bogdan,
thanks for your guide. this solution is dependent to RTPProxy and I think this
should be on the same server with opensips. this thing have any impact on
performance? in this solution, we can play something when we received 180 or
183? and after the call answered we should care about
Hi,
Thank you for your reply. I want do this exactly same as Jayesh said. Jayesh
did you implement this? how?
Stefano, scripting with perl is a good solution with high performance?Have you
any example that help me?
Regards,
H.Yavari
From: Stefano Pisani
To:
Hello,
You can simply do ring back tones with OpenSIPS and RTPProxy - it can
inject media in a call in early stage. See the rtpproxy_stream2uac()
functions:
http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html#rtpproxy_stream2xxx
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Dev
You can develop something using perl and Net::SIP module.
Il 13/02/2014 19.45, Jayesh Nambiar ha scritto:
Hi,
CRBT is caller ring back tone. What you are primarily looking at is
sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you
control both legs of the call. So when you ge
>> What will help you is a kind of route_to_gw() but instead of taking as param
>> a single GW, >> to take a list of GWs ? and to iterate through using
>> use_next_gw() ?
That's exactly what I need! :) Or a do_routing() gw_whitelist which is
executed in order.
As you would imagine we already `us
Hi,
CRBT is caller ring back tone. What you are primarily looking at is sending
the INVITE to some b2bua like FreeSWITCH or Asterisk where you control both
legs of the call. So when you get a ringing signal from the B-leg, you play
some media file on the A-leg.
--- Jayesh
On Friday, February 7, 2
Hello Aki,
Such compressing of the body is not standard (AFAIK, so please correct
me) - so something like this will work only between 2 OpenSIPS (one to
compress, the other to decompress) .Which will tremendously limit
the usability of this feature.
Is there something wrong in what I'm s
Hello all ,
We made a new evaluation of the thinks we plan to have ready for 1.11
and time is not enough. In order to get a valuable and consistent
release, the decision was to delay the 1.11 release for mid March - this
will allow us to complete the ongoing tasks and have a really good set
o
Hello,
FreePBX acts as a B2BUA, so instead of proxying the incoming calls (as
opensips does), it will create a new call (different callid, contact, etc).
In OpenSIPS you can modify the Contact header, but you really have to
understand what you are doing otherwise you will break the routing of
Hi Nick,
What will help you is a kind of route_to_gw() but instead of taking as
param a single GW, to take a list of GWs ? and to iterate through using
use_next_gw() ?
Also, what Nick Altman says is very very true - you can simply populate
the AVPs he mentioned (do it from script, loading fr
Some updates after checking the code - it seems the code allows you do
use it from failure route (this is fixed in all versions starting with
1.8 )...The docs are not updated, but I will take care of that.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solution
Hello,
It seems to be an unfortunate limitation - it should be possible to call
the function from failure route.
I will fix it in the next days - as quick workaround for you , move the
set_rtp_proxy_set() into a separate route and call this route from
failure route - it should do the trick.
Hello Ahsan,
Thank you for the valuable input. I CC'ed here Ovidiu - he is the author
of the QoS module and maybe he can help here.
I see the crash is reproducible - can you do it in a non-production env
where we can enable some more debugging ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founde
Hi all,
On receiving an INVITE, I invoke set_rtp_proxy_set() and rtpproxy_offer() to
request the RTP proxy open ports for RTP packets relay. When the callee
rejects the INVITE and replies with 486 response, I would like to invoke
set_rtp_proxy_set() and unforce_rtp_proxy() to notify the RTP proxy
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