Hi All,
I wonder if any of you has some suggestions on which WebRTC
client/softphone to use for a click-to-dial, webpage hosted solution. Any
suggestions?
Thanks
l.
--
Loway - home of QueueMetrics - http://queuemetrics.com
Test-drive WombatDialer beta @ http://wombatdialer.com
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Matthew,
Yes that's correct, when I use u-law call works fine.
In case of g729, I enabled sip debug with 'sip set debug on' and captured all
the sip traces and got whatever I posted in last email. There was no other call
on the system when I captured sip trace. Please suggest further
Kamlesh Kumar wrote:
Yes that's correct, when I use u-law call works fine.
In case of g729, I enabled sip debug with 'sip set debug on' and captured all
the sip traces and got whatever I posted in last email. There was no other
call on the system when I captured sip trace. Please suggest
thanks justin i try to do this but the issue still the same.this link is
stored in my server 192.168.5.109 .but what i want to receive this link
when i call this number in my pc
ip adresse of my pc 192.168.5.131
ip adresse of server when the page php is stored
thanks and regards
2013/5/30
El 31/05/13 09:21, Salaheddine Elharit escribió:
thanks justin i try to do this but the issue still the same.this link is stored
in my server 192.168.5.109 .but what i want to receive this link when i call
this number in my pc
ip adresse of my pc 192.168.5.131
ip adresse of server when the
hello ,
thanks alex for your help and support the scenario is correct.
i will try to follow your suggestion and i will update you asap
thank you again for your explication i really appreciate it
2013/5/31 Alex Villacís Lasso a_villa...@palosanto.com
El 31/05/13 09:21, Salaheddine Elharit
On Friday 31 May 2013, Alex Villacís Lasso wrote:
From this discussion, I am guessing the following scenario. Please correct
me if I am wrong. - There are (at least) three roles in your scenario: the
Asterisk server, the PHP webserver (which may or may not be the same
machine as the Asterisk
On Fri, May 31, 2013 at 11:29 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hello ,
thanks alex for your help and support the scenario is correct.
i will try to follow your suggestion and i will update you asap
thank you again for your explication i really appreciate it
I was wondering if there is anyone following the list that has this or
similar PCI cards running in a 1U server. If so, could you possibly
recommend a PCI card bus adapter? I'm not sure exactly which kind I'll
need.
I plan on researching the different types of adapters I can purchase for
my
I have some 1-U servers running using Xeon processors with the Asus
RS100-E7/PI2 barebone. There's only room for a single card and in my
case I am using PCI-E Sangoma B500 cards. Geometry is the basic
problem. But this should not be a problem for your T1/E1 card. I had
problems using a
On 05/31/2013 03:42 PM, jg wrote:
I have some 1-U servers running using Xeon processors with the Asus
RS100-E7/PI2 barebone. There's only room for a single card and in my
case I am using PCI-E Sangoma B500 cards. Geometry is the basic
problem. But this should not be a problem for your T1/E1
Voxeo/Phono webrtc.
/Adnan
On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
Hi All,
I wonder if any of you has some suggestions on which WebRTC
client/softphone to use for a click-to-dial, webpage hosted solution. Any
suggestions?
Thanks
l.
--
Loway - home of
OK, I need a bit of help here. I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules drop for a day or so.
When I logged in today, I found messages like the ones below on my
asterisk console. Obviously somebody was trying to take advantage of
my carelessness. So can
... an anonyous (not registerted) sip user from 188.161.238.232 was
trying to initiate a call to
9725955 and so on...
you could enable sip tracing to get more information.
maybe you should change the 'allowguest' option in sip.conf..?
regards,
yves
Am 31.05.2013 23:57, schrieb Chris Gentle:
Top of sip.conf
quote
;
; SIP Configuration example for Asterisk
;
; Note: Please read the security documentation for Asterisk in order to
; understand the risks of installing Asterisk with the sample
; configuration. If your Asterisk is installed on a public
; IP address
OK, I understand now. I didn't realize allowguest was on by default.
I guess I should read more closely. Thanks!
On Fri, May 31, 2013 at 5:15 PM, Yves A. yves...@gmx.de wrote:
... an anonyous (not registerted) sip user from 188.161.238.232 was trying
to initiate a call to
9725955 and so
Hello;
When I type make menuselect and finding the channels that has the sign XXX
before it (this at the driver), how can I know the dependencies that are
causing this conflict?
Regards
Bilal
--
_
-- Bandwidth and Colocation
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Le 31/05/2013 15:10, bilal ghayyad a écrit :
Hello;
When I type make menuselect and finding the channels that has the sign XXX
before it (this at the driver), how can I know the dependencies that are
causing this conflict?
Dependencies are
hello;
hopefully u can help me
i have asterisk vanilla installation 11 and i have also managed to install
webrtc2sip
how do i make asterisk to communicate with webrtc2sip c'se right now both
run independently
Regards.
Kyeyune Bob
Network IT Engineer
+256 774 702 258
bob.kyey...@onesolutions.ug
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