Doesnt reinvite still keeps asterisk in SIP path?
lt looks like all it does is making both peers send rtp to each other.
what i'm trying to do is, a sip peer sends invite to asterisk, asterisk
calls another phone and in case of no answer, "tells peer" to send this call
to another number outside an
Reinvite its called
On Sat, Mar 12, 2011 at 1:22 PM, Al lists wrote:
> is there a way to have asterisk to flash transfer the call and not being in
> sip path anymore?
> for example, a sip trunk send a call to asterisk, asterisk rings a handset,
> then sends/flash hooks the call to a cell phone th
Hi ,
telco=>asteriskserver=>eppbx
I had two 4 span cards in server , one 4span connected from telco and
another one is connected to eppbx.
and when i am run the dahdi_tool commad it was showing like spna1 to span4
then also span1 to span4 all is showin OK .but i need to show span1 to
span8.
and i
Disclaimer: i took a quick look at wikipedia before replying.
Sept 2009 isn't that old compared to G.711 itself ;)
But yea, I hadn't heard about that extension until today.
-Kyle
On Sat, Mar 12, 2011 at 7:27 AM, Steve Underwood wrote:
> Hi,
>
> Has anyone seen G.711.0 in real world use? The spe
is there a way to have asterisk to flash transfer the call and not being in
sip path anymore?
for example, a sip trunk send a call to asterisk, asterisk rings a handset,
then sends/flash hooks the call to a cell phone through sip trunk, and not
being in path anymore?
--
Not sure if this is appropriate but I know it's tough out there right
now. We're hiring a full-time linux/asterisk tech in the N. Metro
Atlanta area. For more info contact mignacio at voicenation.com
--
_
-- Bandwidth and Coloc
On Sat, 12 Mar 2011, Thomas Winter wrote:
when I audio studio should produce an sound file to play back with
Asterisk. Whats the best format they should deliver the audio file?
I like to receive audio at the highest quality the studio can provide and
then transcode down to what Asterisk can h
AH!! Boy I didn't notice that it's not AGI.
But In 1.2 it's working with AGI apps.
When i am running it on bash it excute successfully and ringing phone
but no auto answer working.
Let me try with system applications and I will let you know.
Thanks for helping me with this.
--
Sent from my
Hi,
Has anyone seen G.711.0 in real world use? The spec was published quite
a while ago, but as far as I can tell there is no RFC defining the SDP
and RTP details needed to deploy it, and nobody advertises that they
support it in their products.
Steve
--
___
Hi,
when I audio studio should produce an sound file to play back with
Asterisk. Whats the best format they should deliver the audio file?
Sample Size: 16-bit (2 bytes)
Sample Encoding: signed (2's complement)
Channels : 1
Sample Rate: 8000
thanks
Thomas
--
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