Hi,
I was testing with newly introduced websocket support in asterisk 11. I
have successfully implemented everything except when i try to make a call i
get no audio. I have tried both SipML5 as well as SIP-JS as clients. the
call get connected but i never hear any audio stream. I however get the
I have tried it on raspberrypi, Although i didn't do any tests but looks
promising. Should be able to handle calls in figure of two digits easily,
The final answer always depends on your configuration.
Regards,
Qasim
On Tue, Sep 4, 2012 at 8:52 AM, Sazzad sazzadbinka...@gmail.com wrote:
has
call recording to be not too much. However I used the packages from the
repo, maybe compiling it yourself and leaving out unnecessary stuff gives
beter performance.
2012/9/4 qasimak...@gmail.com qasimak...@gmail.com
I have tried it on raspberrypi, Although i didn't do any tests but looks
at 2:45 PM, Stefan at WPF
stefan.at@googlemail.comwrote:
Guess this is what most people are doing by compiling only necessary
stuff. Personally I find this is to much fidling and contraproductive. Just
bought a small Atom System. Hope it works better.
2012/9/4 qasimak...@gmail.com qasimak
Thanks :).
Regards,
Qasim
On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen
james.morten...@a-cti.comwrote:
qasimakhan at gmail.com qasimakhan at gmail.com writes:
Hi,I was testing with newly introduced websocket support in asterisk 11.
I
have successfully implemented everything except
SVN Version is always development version. Try downloading a stable tarball
archive from http://www.asterisk.org/downloads.
Regards,
Qasim
On Thu, Nov 1, 2012 at 6:52 PM, Thomas Thomas debussy...@gmail.com wrote:
Hello,
I installed Asterisk 11 via the following command
* svn co
You can also hardcode these values in call.htm find below lines:
i_port = 4062 + (((new Date().getTime()) % 5) * 1000);^M
s_proxy = sipml5.org;^M
and change them to
i_port = port/ws;^M
s_proxy = ws://* server IP:;^M
Change port
You can use Radius Agi developed by PortaOne from following link.
http://www.voip-info.org/wiki/view/PortaOne+Radius+auth
Regards,
Qasim
On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini samiramhosse...@yahoo.com
wrote:
Hi all,
based on the following link, I am going to authenticate SIP
Hi Kamlesh,
Asterisk give you very less control over SIP messaging. You can how ever
add/remove/modify SIP headers from initial invite only. To modify a sip
header you can use asterisk function *SIP_HEADER(name)*. If you want to
permanently change date why not change system date/time?
Regards,
Search jitter in sample sip.conf. Everything is well documented there.
Regards,
Qasim
On Tue, Apr 23, 2013 at 3:03 PM, Muhammad Yousuf muyous...@gmail.comwrote:
I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in
server. I am having some issue in audio quality. I want
Read up on new features and changelog of asterisk 11 you'll find the
changes there.
Regards,
Qasim
On Thu, Apr 25, 2013 at 3:32 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hello;
How I can compare between Asterisk 1.8 and 11 with reference to the
following points:
1) SMS.
2) gtalk and
Hi David,
Direct media should work either way. if your phones are behind NAT you will
also require the NAT option enabled in asterisk, How ever the tricky part
in all this is that you wont be able to acurately keep track of calls on
these phones. If or any unforeseen reason the phone goes
Hi faheem,
You can do this:
ACTION: Redirect
Channel: Channel ID
Context: Context
Exten: Exten
Priority: Priority
Regards,
Qasim
On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote:
Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want
It depends on chan_local see if that is enabled or not.
Regards,
Qasim
On Mon, May 27, 2013 at 11:56 AM, upendra uppi...@gmail.com wrote:
Hi,
i am trying to install asterisk newer version on the Elastix machine, but
while installing the chan_sip,c module is not building while make. when i
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