ch/cloud-client
>
> Hope that all makes sense!
>
> On 19 July 2017 at 10:17, Rahul MathuR <rahul.ultim...@gmail.com> wrote:
> > Hi Jonathan
> >
> > Thanks !
> > That would indeed be wonderful, at this point I really do not care
> whether I
> > nee
oud_api.zip>.
>
> You can also clone the project with Git <https://git-scm.com/> by running:
>
> $ git clone git://github.com/zaf/asterisk-speech-recog
>
>
> On 19 Jul 2017 11:36 am, "Rahul MathuR" <rahul.ultim...@gmail.com> wrote:
>
>> Hi
a service key but I have entered my API key in the script in
> the 'User defined parameters' section. You did that, right? What do the
> other user defined parameters in your script look like?
>
>
> On 7/19/2017 4:37 AM, Rahul MathuR wrote:
>
> Hi,
>
> I'm trying to int
ady tried the cloud_api branch?
>
> Regards,
>
> Marcelo H. Terres <mhter...@gmail.com>
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter.com/mhterres
> https://linkedin.com/in/marceloterres
>
> On 19 July 2017 at
> On 19 July 2017 at 09:37, Rahul MathuR <rahul.ultim...@gmail.com> wrote:
>
>> Hi,
>>
>> I'm trying to integrate Google cloud speech recognition v2 in it. I can
>> get the audio recorded, have created Service key and API key but whenever I
>> try to access
Hi,
I'm trying to integrate Google cloud speech recognition v2 in it. I can get
the audio recorded, have created Service key and API key but whenever I try
to access it, I just get 403 access denied. I am at my wits end here.
Has anybody tried it ? were you successful ? Could you please guide me
Hi guys,
Could you please let me know whether the latest Asterisk has a support for
inbound UPDATE ?
In my case, the carrier is sending an UPDATE to change the codec which is
replied by 5xx from Asterisk 11.17.1.
Thanks.
--
_
Hello Russell,
I meant, 408 Request Time-out.
On Thu, May 21, 2015 at 11:33 PM, Russell Treleaven rtrelea...@bunnykick.ca
wrote:
what is RTO'ed.?
On Thu, May 21, 2015 at 12:50 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello folks,
Can somebody please guide me on this.
Thanks
,
Russell
On Thu, May 21, 2015 at 2:17 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello Russell,
I meant, 408 Request Time-out.
On Thu, May 21, 2015 at 11:33 PM, Russell Treleaven
rtrelea...@bunnykick.ca wrote:
what is RTO'ed.?
On Thu, May 21, 2015 at 12:50 PM, Rahul MathuR
Hello folks,
Can somebody please guide me on this.
Thanks !
On Wed, May 20, 2015 at 7:42 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello,
I'm facing issues with far-end NAT traversal this case being, (wifi to 3G)
is still not working.
I am using Linphone-Android, one on 3G
, 2015 at 7:26 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello Vasiliy,
Thanks for replying.
Not sure why Linphone-Android won't receive INVITE, since it is responding
well to the keep-alive OPTIONS messages from proxy.
Both, proxy and SIP server are sending packets to UAb on which UAb
Hello,
I'm facing issues with far-end NAT traversal this case being, (wifi to 3G)
is still not working.
I am using Linphone-Android, one on 3G and other behind Wifi router.
I am able to call from 3G to Wifi, Wifi to Wifi (using same Wifi router)
but for some reason, Wifi to 3G is constantly
Hello Vasiliy,
Thanks for replying.
Not sure why Linphone-Android won't receive INVITE, since it is responding
well to the keep-alive OPTIONS messages from proxy.
Both, proxy and SIP server are sending packets to UAb on which UAb is
apparently responding to only proxy.
Is this a genuine flaw in
Hello Manuel,
To support the hypothesis of crypt libs screwing the logic, you can try a
'secure call' without using webrtc.
If them are to be blamed; your 'secure call' won't be successful.
Aside this, you can get a better idea of what has dwell-ed behind the
curtains by looking at syslogs.
are the way that we detect a
video call, how we route to our backend servers, and that we send video
calls directly to a registered peer and not the the backend Asterisk
servers.
On Thu, Feb 12, 2015 at 12:34 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Gentle Reminder !
Thanks
Warm
Gentle Reminder !
Thanks
Warm Regds,
Rahul
On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Thanks guys !
I did further investigation of the Chrome logs and found that... (this is
really interesting), even though I disabled Video; still JSsip was sending
to resolve
itself.
Hope that helps,
Marc
On Tue, Feb 10, 2015 at 12:01 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello gents,
I was trying my hands on getting a successful RTCweb call (JSsip, since
Peter Dunkley mentioned that he's been using JSsip for most of the testing
scenarios
Hello,
I was wondering whether Kamailio (as proxy) can generate a PRACK on its own
( since one of the custom written dialer is not sending PRACK) ?
Is there any way I can achieve this ?
--
Warm Regds.
MathuRahul
___
SIP Express Router (SER) and
, 2015 at 2:14 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfu...@sipwise.com wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
Hello,
I am totally
: sipml5 -- ws/wss -- Ec2 Kamailio --sip udp-- FS --sip
udp-- *
media: sipml5
*
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hi Richard,
Thanks for spending some cycles
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where the
SIP server sends 183 session in progress to kamailio but after that I can
only see
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where the
SIP server sends 183 session in progress to kamailio but after that I can
only see
-route for rr module ).
My 2 cents.
Le 26/01/2015 11:21, Rahul MathuR a écrit :
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfu...@sipwise.com wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
Hello,
I am totally struck at a point while implementing Kamailio as proxy
stable version from 4.2 branch, respectively
4.2.1 at this time -- you are running a pre-release version, which was not
supposed to be ready for production anyhow, and there was a fix to this
issue already.
Cheers,
Daniel
On 28/11/14 19:59, Rahul MathuR wrote:
Hello,
I have recently
Hello,
I have recently moved some traffic to kamailio-4.2 (installed on a seperate
box) and it crashes quite often. Below is the backtrace of the core files -
CORE#1
(gdb) bt
#0 0x0033d8c32635 in raise () from /lib64/libc.so.6
#1 0x0033d8c33e15 in abort () from /lib64/libc.so.6
#2
Hello Dev,
I have recently moved some traffic to kamailio-4.2 (installed on a seperate
box) and it crashes quite often. Below is the backtrace of the core files -
CORE#1
(gdb) bt
#0 0x0033d8c32635 in raise () from /lib64/libc.so.6
#1 0x0033d8c33e15 in abort () from /lib64/libc.so.6
#2
Hello,
I'm new to WebRTC although I've been using kamailio as sip proxy server for
few months now. What I really do not know and trying to understand is -
a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to
plain UAC/WebRTC based UAC ?
b) What to use for media proxying
Hello,
We have observed a strange behavior in corex module that it gets loaded at
every sip packet which arrives to kamailio.
We put a static variable and saw that it gets re-initialized to 0 everytime
any sip packet comes to it.
Could you please tell me how to stop it and load it just once.
Hello,
We have observed a strange behavior in corex module that it gets loaded at
every sip packet which arrives to kamailio.
We put a static variable and saw that it gets re-initialized to 0 everytime
any sip packet comes to it.
Could you please tell me how to stop it and load it just once.
/corex.html#idp29928
If you interconnect with other servers/gateways when you don't what to do
special encoding, then you need to test src ip or look ar r-uri/dst uri.
Cheers,
Daniel
On 17/09/14 04:24, Rahul MathuR wrote:
Hi,
Did you get some free cycles to look at it ?
On Wed, Sep 17
Hello,
I was going through the new features and stumbled upon this new one -
developed by Mohd. Shahzad Shafi.
As already mentioned on the wiki about this module, I intend to use it for
my custom security layer between UACs and SIP Proxy (Kamailio) but the
issue is - the custom security layer
choice as well,
e.g. Python, LUA, JAVA and so on.
I would recommend the second option, as it has less processing overhead
for kamailio.
Thank you.
On Tue, Sep 16, 2014 at 6:09 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello,
I was going through the new features and stumbled upon
Hi,
Did you get some free cycles to look at it ?
On Wed, Sep 17, 2014 at 12:12 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Thanks for replying !
But how to check whether a particular message received by Kamailio was
sent by UAC or SIP Server ?
Also, on the same lines - how to know
Hello,
I was going through the new features and stumbled upon this new one -
developed by Mohd. Shahzad Shafi.
As already mentioned on the wiki about this module, I intend to use it for
my custom security layer between UACs and SIP Proxy (Kamailio) but the
issue is - the custom security layer
choice as well,
e.g. Python, LUA, JAVA and so on.
I would recommend the second option, as it has less processing overhead
for kamailio.
Thank you.
On Tue, Sep 16, 2014 at 6:09 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello,
I was going through the new features and stumbled upon
Hi,
Did you get some free cycles to look at it ?
On Wed, Sep 17, 2014 at 12:12 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Thanks for replying !
But how to check whether a particular message received by Kamailio was
sent by UAC or SIP Server ?
Also, on the same lines - how to know
.
Op 12-aug.-2014, om 14:39 heeft Rahul MathuR rahul.ultim...@gmail.com
het volgende geschreven:
Hello Davy,
Thanks for writing back..
Tonight I'll take the tcpdump on Kamailio box and share the file.
Please note that Kamailio and Freeswitch are both on public IP at
Freeswitch param
:39 heeft Rahul MathuR rahul.ultim...@gmail.com
het volgende geschreven:
Hello Davy,
Thanks for writing back..
Tonight I'll take the tcpdump on Kamailio box and share the file.
Please note that Kamailio and Freeswitch are both on public IP at
Freeswitch param, enable_timer=false is set
Thanks for the help !!
It is now resolved.. my special thanks to Davy !!
On Thu, Aug 14, 2014 at 7:12 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello Davy,
My constraint is that I cannot use TCP for this solution.
I am attaching my kamailio.cfg file, please please help me
Hello,
I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and Sip
server FreeSwitch.
Whenever I call directly from UAC to Sip server, the call gets established
for as long as I want, however when I use the proxy in between, it gets
disconnected within 30 seconds. It seems that FS
the
call has failed.
Do you have a trace of the packets?
grtz,
Davy Van De Moere
2014-08-12 13:37 GMT+02:00 Rahul MathuR rahul.ultim...@gmail.com:
Hello,
I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and
Sip server FreeSwitch.
Whenever I call directly from UAC to Sip
Hello,
I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and Sip
server FreeSwitch.
Whenever I call directly from UAC to Sip server, the call gets established
for as long as I want, however when I use the proxy in between, it gets
disconnected within 30 seconds. It seems that FS
Thank you Victor !
On Fri, Aug 1, 2014 at 8:03 PM, Victor Seva
linuxman...@torreviejawireless.org wrote:
http://www.asipto.com/pub/kamailio-devel-guide/
On 1 Aug 2014 19:09, Rahul MathuR rahul.ultim...@gmail.com wrote:
Hello,
Could anybody please guide me to some documentation available
Hello,
Could anybody please guide me to some documentation available for writing a
new module for kamailio. I am also looking to tap the data coming on its
sockets both TCP UDP.
Thanks in advance !
--
Warm Regds.
MathuRahul
___
sr-dev mailing list
On Thu, Jun 5, 2014 at 8:28 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello Carlos,
Did you get a chance to look at the attached configuration file ?
Warm Regds,
Rahul Mathur
On Thu, Jun 5, 2014 at 2:46 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hi Carlos,
Please find
Hello Carlos,
Did you get a chance to look at the attached configuration file ?
Warm Regds,
Rahul Mathur
On Thu, Jun 5, 2014 at 2:46 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hi Carlos,
Please find attached the configuration file which I am using.
Kindly refer route[CNXCC] where I
.
This will of course lead to 1 call being reported always, since
request-URIs tend to be different from each other.
Regards,
Carlos
On Tue, Jun 3, 2014 at 7:01 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hi Carlos,
Sorry for coming late on this one.
the ruri value in my example
, 2014 at 7:52 PM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Further, from the logs it seems that cnxcc_set_max_channels is returning
-1 and I have no idea why it is failing.
On Sat, May 17, 2014 at 5:48 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello Carlos,
Many thanks for your
are active before doing the restart. This
is only true for cnxcc, and not for dialog module for example, which has
the option of backing everything up to a permanent storage such a database.
Regards,
On Tue, May 27, 2014 at 4:39 AM, Rahul MathuR rahul.ultim...@gmail.com
wrote:
Hello
Hello,
Is it possible to reload kamailio.cfg at the runtime after doing few
modifications on the business rules laid under CNXCC module.
Eg:
#!ifdef CNXCC_CHANNEL
xlog(L_INFO, Setting up channel based credit control);
$var(max_chan) = 2;
$var(retcode) = cnxcc_set_max_channels($var(client),
transactions, they typically
recover due to retransmissions.
Cheers,
Daniel
On 27/05/14 09:48, Rahul MathuR wrote:
Hello,
Is it possible to reload kamailio.cfg at the runtime after doing few
modifications on the business rules laid under CNXCC module.
Eg:
#!ifdef CNXCC_CHANNEL
Hello,
Could somebody please help me out, I am a newbie to kamailio world !
Thanking you in anticipation ...
On Thu, May 15, 2014 at 1:52 PM, Rahul MathuR rahul.ultim...@gmail.comwrote:
Thanks Juha Daniel for the quick reply !
But is there any example which shows how can I create
*Hello,*
*Please accept my apologies for asking a simple question - Is there
any example which shows how can I create the dialog profile **value
based on the SIP domain or SIP Server IP (IP like- 112.23.134.5). I**
need to control many SIP Server IPs with limited channels to each with
my**
/docs/modules/stable/modules/cnxcc.html#idp132608
On Fri, May 16, 2014 at 3:28 PM, Rahul MathuR rahul.ultim...@gmail.comwrote:
*Hello,*
*Please accept my apologies for asking a simple question - Is there any
example which shows how can I create the dialog profile **value based on the
SIP
of the request-uri, $rd.
Regards,
Carlos
On Fri, May 16, 2014 at 4:28 PM, Rahul MathuR rahul.ultim...@gmail.comwrote:
Thank you very much Carlos !
One last thing, can I use $fd in cnxcc_set_max_channels as below -
cnxcc_set_max_channels($fd, $var(max_chan));
to restrict the simultaneous calls
Further, from the logs it seems that cnxcc_set_max_channels is returning -1
and I have no idea why it is failing.
On Sat, May 17, 2014 at 5:48 AM, Rahul MathuR rahul.ultim...@gmail.comwrote:
Hello Carlos,
Many thanks for your help.
I followed the steps you mentioned for CNXCC. But even
, 2014 at 7:55 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
dialog module can be used for limiting number of active calls -- read
about profiles in dialog module readme.
Cheers,
Daniel
On 14/05/14 20:37, Juha Heinanen wrote:
Rahul MathuR writes:
So basically, n UACs
Hello,
I've run into a situation where I want to throttle the number concurrent
calls proxied by Kamailio based upon the SIP Server through which they are
intended to go to the end point.
The network diagram is somewhat like -
UACs (n) --- Kamailio (on public IP) -- SIP Servers (m)
, Rahul Mathur sriv...@gmail.com wrote:
I am newbie and looking for Qt support with ZeroMQ on Linux for desktop
application.
Can I have some links to perform below -
1. Qt bindings support to ZeroMQ
2. Installation
3. Executing ZeroMQ PUB-SUB with Qt enabled user inputs on it's GUI
All,
Newbie to Qt development and it's usages. Need some help to KICK START Qt
usages.
I am using Linux desktop platform. I have following futures to implement -
1. A GUI to accept some inputs through multiple buttons.
2. Above values should be taken either as INPUT or LIMIT set for any data
or
All,
Newbie to Qt development and it's usages. Need some help to KICK START Qt
usages.
I am using Linux desktop platform. I have following futures to implement -
1. A GUI to accept some inputs through multiple buttons.
2. Above values should be taken either as INPUT or LIMIT set for any data
or
All,
Looking if ZeroMQ based code can be profiled using Intel VTune profiler or
any other profilers.
What are the options to set to profile ZeroMQ based code?
Thanks
___
zeromq-dev mailing list
zeromq-dev@lists.zeromq.org
All,
Below code is working fine when SSOMSSub () and SSOMSPub () API are written
as separate two files (sub.cpp and pub.cpp) but it doesn't execute properly
when both API SSOMSSub () SSOMSPub () are written in same single
test1.hpp file as below and this file being called by driver file test.cpp
All,
I have testpub.cpp and testsub.cpp as below -
---testpub.cpp---
#include cstdio
#include cstdlib
#include string
#include zmq.hpp
#include zmq_utils.h
using namespace std;
struct MessageStruct {
longUniqueID;
short Pro_ClientIndicator;
char
Hi,
I am using From and To datpickers, when I select date from from
datepicker, I want all dates in calender before 'From date' to get
disabled. Like if, 'From date' chosen from it is : 22/12/2011, I want
all dates before it in calendar to be seen as disable.
Please reply with your valuable
hello All,
Could anyone guide me in implementing SNOOP n I-TCP..
Thanks in advance
Rahul
I m new to NS2. Could anyone tell me how to implement Snoop , WTCP and I-TCP
in NS2.
Thanks in advance.
Regards,
Rahul Mathur
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