Displaying 8 results from an estimated 8 matches for "deltapath".
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datapath
2004 Aug 25
0
How to be taken out of the queue?
...he context? Any reply would be much
appreciated!
A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
context = qoutcon
Kenny Lam
SIP Application Engineer
Deltapath Commerce & Technology Limited
---------------------------------------
SIP By Deltapath!
www.deltapath.com
2004 Jan 17
1
Voicetronix OpenLine4: disable answering on a particular channel & delay before dial
2004 Dec 27
2
MYSQL_FRIENDS
Hello *'s,
Hi, I've just tried to enable MYSQL Friends in CVS HEAD. But i cannot
find this option.On wiki i found this.
To enable this, you need to edit the Makefile in the channels directory
of your source tree and enable MYSQL_FRIENDS. This enables database
definition of both IAX2 and SIP friends. Make sure you have the MySQL
development kit (libraries) installed before
2005 Jan 08
1
What is acceptable network latency for voipconnection?
...> David
>
>
>
> On Sat, 8 Jan 2005 06:22:58 -0800 (PST), Robert Augustyn wrote
> > Very good point.
> > So what can you do ( if anything ) to control the load
> > on the network outside of your control?
> > robert
> >
> > --- David Liu <david@deltapath.com> wrote:
> >
> > > Assuming the network loading is fairly constant,
> > > 300ms latency is actually not
> > > noticeable unless you put both phones next to your
> > > ears to compare.
> > >
> > > Latency affects delay while network l...
2005 Jan 31
3
Announcement to caller when called party haspicked up - without initial Answer()?
> -----Original Message-----
> From: David Liu [mailto:david@deltapath.com]
> Sent: 31 January 2005 14:34
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Announcement to caller when
> called party haspicked up - without initial Answer()?
>
>
> This is super easy to do. All you need to do is to put...
2005 Jan 09
2
What is acceptable network latency forvoipconnection?
...t
> > Augustyn wrote
> > > > Very good point.
> > > > So what can you do ( if anything ) to control
> > the load
> > > > on the network outside of your control?
> > > > robert
> > > >
> > > > --- David Liu <david@deltapath.com> wrote:
> > > >
> > > > > Assuming the network loading is fairly
> > constant,
> > > > > 300ms latency is actually not
> > > > > noticeable unless you put both phones next to
> > your
> > > > > ears to compar...
2006 Feb 21
0
Session Media 183 and Ringing Tone 180 Passing To SIP At the Same Time
Hi there,
I am seeing some very interesting thing with the latest Zaptel 1.2.X, hope may
be someone can shed some light on this.
Normally, to dial via your Zaptel T1 card, you would do something like:
;Dial to PSTN
exten => _9.,1,Dial(Zap/g1d/{EXTEN:1})
by not adding any option after the extension e.g. no "r" and no "m", Asterisk
will pass thru the session media from the
2005 Jan 31
5
Announcement to caller when called party has picked up - without initial Answer()?
This is super easy to do. All you need to do is to put that announcement in
a MP3 and set a musiconhold class for that incoming Zap channel. Then
basically when ever that PSTN number rings, Asterisk will play the MP3
stream "Your call may be monitored or recorded, please hangup if you do not
agree...etc" in a loop until the line is answered. Caller doesn't pay a
single dime to