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 proceed
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 sugg
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 Just
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 pag
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
> El 31/05/13 09:21, Salaheddine Elharit escribió:
>
> thanks ju
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 Aster
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 i
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 spec
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 Sangoma
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/E
Voxeo/Phono webrtc.
/Adnan
On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri 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 QueueMetric
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 s
... 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:
O
Top of sip.conf
;
; 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 connected to
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. wrote:
> ... an anonyous (not registerted) sip user from 188.161.238.232 was trying
> to initiate a call to
> 9725955 and so on...
> you coul
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
--
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-- Bandwidth and Colocation P
-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 ar
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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