stop recording when there is 5
seconds of silence (which is what I set).
Is there a setting somewhere that I'm missing somewhere for a silence
threshold for WaitForSilence or am I misunderstanding its use?
The Asterisk vers
stop recording when there is 5
seconds of silence (which is what I set).
Is there a setting somewhere that I'm missing somewhere for a silence
threshold for WaitForSilence or am I misunderstanding its use?
The Asterisk vers
wme/sorry
[mleonetti]
context=internalphones
number=>xx,20
Obviously I omitted my cell phone number.
I don't see anything crazy when attaching to asterisk.
--
Mike A. Leonetti
As warm as green tea
--
_
-
Is it possible to have Asterisk resend the SIP credentials in every INVITE?
--
Mike A. Leonetti
As warm as green tea
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They are saying that the Asterisk registration doesn't have an
"Allow-Events" and an "Allow" in the header. Would this cause
On 05/07/10 12:40, Gareth Blades wrote:
> Mike A. Leonetti wrote:
>
>> On 05/07/10 12:14, Gareth Blades wrote:
>>
>>> Mike A. Leonetti wrote:
>>>
>>>
>>>> On 05/07/10 11:52, Gareth Blades wr
On 05/07/10 12:40, Gareth Blades wrote:
> Mike A. Leonetti wrote:
>
>> On 05/07/10 12:14, Gareth Blades wrote:
>>
>>> Mike A. Leonetti wrote:
>>>
>>>
>>>> On 05/07/10 11:52, Gareth Blades wr
On 05/07/10 12:14, Gareth Blades wrote:
> Mike A. Leonetti wrote:
>
>> On 05/07/10 11:52, Gareth Blades wrote:
>>
>>> Mike A. Leonetti wrote:
>>>
>>>
>>>> In an attempt to connect our Asterisk 1.6 phone system with ano
On 05/07/10 11:52, Gareth Blades wrote:
> Mike A. Leonetti wrote:
>
>> In an attempt to connect our Asterisk 1.6 phone system with another
>> phone system called "Broadsmart", they gave me credentials to register to.
>>
>> Connected to Asterisk 1.6.
rt,706)
and the Macro
[macro-broadsmart]
exten => s,1,Dial(SIP/${ar...@broadsmart,60)
Asterisk reports:
[May 7 11:34:45] WARNING[10402]: chan_sip.c:17775
handle_response_invite: Received response: "Forbidden" from '"Mike A.
Leonetti" ;tag=as6376d669'
The peop
Ever since the upgrade to Asterisk 1.6 the e-mails from Asterisk are
coming from r...@.
In the voicemail.conf I have
fromstring=Asterisk PBX
serveremail=asterisk
And in my ssmtp.conf
root=asterisk
However they still come from root@
--
___
Michael Leonetti wrote:
> I have an a bunch of SPA941 Linksys phones for users in and out of the
> office. When the phones are in the office (and on the same network as
> the asterisk server) the WMI goes on when it should and off when it
> should. But when the phone is on another network and
Sean Brady wrote:
>> To get MeetMe working properly, I know some sort of timing device
>> provided by the zaptel package is required (even if it means the
>> zt_dummy). But, on a virtual machine I know that the Linux timing won't
>> work as expected. Is it possible to then dedicate a physical dev
David Backeberg wrote:
> On Fri, Feb 19, 2010 at 11:42 AM, Mike A. Leonetti
> wrote:
>
>> To get MeetMe working properly, I know some sort of timing device
>> provided by the zaptel package is required (even if it means the
>> zt_dummy). But, on a virtual machine I
To get MeetMe working properly, I know some sort of timing device
provided by the zaptel package is required (even if it means the
zt_dummy). But, on a virtual machine I know that the Linux timing won't
work as expected. Is it possible to then dedicate a physical device
like a USB port or somethi
Perfect. Thanks.
Mike A. Leonetti
As warm as green tea
Evolution CE
3468C Lawson Boulevard
Oceanside, NY 11572
www.evolutionce.com
516-536-5006 ext 105
516-208-4679 (Direct)
Danny Nicholas wrote:
> Set his call-limit to 1 in users.conf. Other than that, you could check the
> channel
A little bit of a strange request.
Basically I want all calls that go to one user go to voicemail
immediately if the user is on the phone. The user is using the Linsys
SPA941, and even though he can be on the phone, calls will still ring
his phone. I tried disabling the rest of the "lines" on th
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