dear
is there any study material for implementing click to call in asterisk.
plz help.
thanks
regards
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10 jul 2012 kl. 20:50 skrev Kevin P. Fleming:
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug that if you have multiple interfaces with
11 jul 2012 kl. 00:26 skrev James Lamanna:
On Mon, Jul 2, 2012 at 12:13 AM, Olle E. Johansson o...@edvina.net wrote:
No.
This is probably because you are using phone numbers as names of devices
with type=friend in sip.conf.
That's generally a bad idea.
The SIP channel matches an
This capability is implanted in Vtiger CRM and some other packages. If you
wanted to do it in a stand-alone fashion, it's a relatively simple task.
I did it in PERL using the Asterisk::Manager package. AFAIK there are PHP
equivalents for this as well.
From:
On the subject of click to call - admittedly not necessarily what the OP
was after - I had some marketing blurb from VMware about Zimbra 8 this
morning. Apparently one of the new shiny features is integrated C2C (and
other unified comms stuff).
Has anyone had a chance to play with the SDK as
On Wednesday 11 July 2012, alok srivastava wrote:
dear
is there any study material for implementing click to call in asterisk.
plz help.
thanks
regards
Dead simple! You need to install Apache on the Asterisk server if you haven't
already. Then use a CGI script like this;
On 12-07-11 10:46 AM, A J Stiles wrote:
Then a GET request to /cgi-bin/place_call?tel=018118055ext=101 will
place a call from extension 101 to telephone number 018118055 in
context outgoing.
Hopefully it doesn't need to be said, but if you are going to put this
solution in place, please
Hi
I'm using asterisk 1.8.7
My dialplan for an inbound number is along the lines of
[default]
exten = idenfier,1,Goto(specific-context,s,1)
[specific-context]
exten = s,1,NoOp()
exten = s,2,Dial(SIP/some-extenion,20)
I have been testing the following 2 scenarios:
1) Caller calls in to
On 07/11/2012 07:51 AM, Olle E. Johansson wrote:
10 jul 2012 kl. 20:50 skrev Kevin P. Fleming:
On 07/10/2012 03:24 AM, Olle E. Johansson wrote:
The Asterisk SIP channel has no knowledge about interfaces and can't
bind to a specific interface for communication. In fact, it's a well known
bug
On Wednesday 11 July 2012, Mike wrote:
On 12-07-11 10:46 AM, A J Stiles wrote:
Then a GET request to /cgi-bin/place_call?tel=018118055ext=101 will
place a call from extension 101 to telephone number 018118055 in
context outgoing.
Hopefully it doesn't need to be said, but if you are going
On 12-07-11 11:50 AM, A J Stiles wrote:
Yes indeed. Note the 192.168 address I used in my other example --
the Asterisk server here is on the LAN side of the router, and there
is no firewall rule anywhere forwarding to its port 80. If for some
reason you have to run Asterisk on a box facing
Fine, did you read the question well and understand about what I am asking?
I know well what Verbose do and what Goto do, and my question is not related to
what they are doing because I used Goto 100 times or more. I have been working
on Asterisk more than 5 years and installed alot of sites.
On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Fine, did you read the question well and understand about what I am asking?
Perhaps I did not understand what you were asking. I thought you were
wanting to do something custom per extension (in the case of my example,
I am using CHANNEL function to store the rtt as variable in cdr.When i
checked the records i found that rtt was always zero.To double check i
turned on rtcp debug I was only able to see sent rtcp,sending rtcp but no
got rtcp.
Also the timestamps in rtp debug were oddly dissimilar.Is there a prblem
I've seen similar.
We tried 4 interfaces. On 4 lans, are these considered to be overlapping?
192.168.1.1
192.168.2.1
192.168.3.1
192.168.4.1
I tried this months ago on 1.8, and set provisioned phones to register with
asterisk interface on the lan they were on.
The phones won't register, we're
Slightly off topic but I have an Asterisk server with Audiocodes HD310
phones, they are running 1.2.2 software and I have 1.6.2 - they come as img
files , but I cannot work out how to load the img file into the phones -
any one know?
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On Thu, 2012-07-12 at 15:49 +1200, Alec Davis wrote:
I've seen similar.
We tried 4 interfaces. On 4 lans, are these considered to be overlapping?
192.168.1.1
192.168.2.1
192.168.3.1
192.168.4.1
Depends on the netmask you use :) Assuming you used /24, so no, they
don't overlap.
This
Similar problem
On 12/07/2012, at 4:36 PM, Jeff LaCoursiere wrote:
On Thu, 2012-07-12 at 15:49 +1200, Alec Davis wrote:
I've seen similar.
We tried 4 interfaces. On 4 lans, are these considered to be overlapping?
192.168.1.1
192.168.2.1
192.168.3.1
192.168.4.1
Running openvpn on
See
Route-Permissions module,
It lets you restrict certain phones/extensions to follow a dial-plan
pattern and dial out to the defined trunk etc meanwhile not breaking any
other functionality or features of FPBX- though you can restrict the
features from this too.
You can also specify routes with an callerid qualifier as 09XX/20X
This would only have it apply to extensions in the 200-209 range
That route can then point to a trunk going nowhere if you want to block them
In freepbx there is a field in outbound route page to select callerid that the
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