On Tue, 2003-08-12 at 11:45, WipeOut . wrote:
> The Cisco is from what I have heard a good phone but is VERY expenisve..
>
> My suggestions would be to go with either a SNOM 200 or a Grandstream Bugetone..
Where can one get a SNOM 200 for less than a Cisco 7960? The Cisco's
are about $300 on eBa
On Fri, 8 Aug 2003, Adams, Gavin wrote:
> Also, we decided to go with actual extension numbers on the phones
> instead of usernames per extension. On the Cisco phones, is there a way
> to change the name/number on the top line (white text on black) to the
> user's name, while having the extension
With the increased traffic as of late, I'm wondering if it is time to
split the list again. Specifically I am wondering if it should be split
along the various VoIP protocols and zap hardware, then leave a general
list that does configuration other than VoIP related?
The hope is that those asking
On Sun, 2003-08-10 at 21:31, Steven Critchfield wrote:
> On Sun, 2003-08-10 at 21:25, Andy Hester wrote:
> > Perhaps there is another way to cut down on increased traffic...
> >
> > Specifically, I would go back to the suggestion of a collaborative website
> > for documentation. Collecting info a
I think we are starting to see what type of logic people are wanting in
fewestcalls and leastrecent strategy.
bkw
On Sun, 10 Aug 2003, Richard Lyman wrote:
> i disagree, instead of thinking 'fallback' how about 'order' the agents
> (by effecting the 'metric') so you 'target' the agent you want f
> From: Martin Pycko [mailto:[EMAIL PROTECTED]
>
> On analog ports you need to Answer
> Ringing
> Wait,2
> and then do something .
>
> That should detect faxes.
Martin,
Could you show an example of this for an incoming PRI -> FXS analog fax?
Regards,
--- Gavin
On Tue, 2003-08-12 at 14:41, Brenton D. Rothchild wrote:
> The only thing I would stress is that if your phone is malfunctioning and
> drawing
> more current than the previous 400mA adapter could source (hence the failure
> of
> the previous adaptor), this larger current adaptor would allow the ph
On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote:
> On Mon, 11 Aug 2003 15:15:08 -0500
>
> Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> > On Monday 11 August 2003 02:46 pm, Eduardo Goncalves wrote:
> > > I've tested with 3 diferent machines. Asterisk didn't detect
> > > them.
> >
> > What
Title: RE: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
Most idle or longest idle should have nothing to do with how long your last call was, or total call time. Longest idle is supposed to be the agent who has been sitting there the longest since the last call was taken.
No
Hi,
Iain Stevenson wrote:
>
> It "should" work with the standard PSTN but you can get problems if you
> connect a X100P to a PBX or ISDN TA. Try editing the wcfxo.c file and
> enabling support for ZERO_BATT_RING (uncomment the #define) then rebuild
> and reinstall the zaptel modules - you will n
On Tue, 2003-08-12 at 11:26, Jamie Carl wrote:
> *This message was transferred with a trial version of CommuniGate(tm) Pro*
> Ooh, can i answer this one? Please??
>
>
> RTFM! :)
>
> http://www.digium.com/handbook-draft.pdf
>
> hehe...
>
What actually _read_ a manual, only wimps do that. Easie
I was trying to do a little searching to see if there has even been a
comparison between Asterisk and VOCAL or any of the other OSS packages?
"Practical Voice Over IP using VOCAL" published by O'Reilly and
Associates, attempts to make a strong case about how scalable VOCAL. Of
course, considering t
Martin,
With KFLAGS+=-DNO_CALIBRATION uncommented, the module loads fine. Without the
calibration.
But I have no dial-tone on port 4. All the three other ports works fine.
Could it be a hardware problem?
Thanks in Advance
Eduardo
On Fri, 1 Aug 2003 16:11:12 -0500 (CDT)
On Mon, 2003-08-11 at 15:53, Mark Spencer wrote:
> > I just hope that the price difference is small enough for Digium to
> > consider this other chipset. From the lists it is obvious that there is
> > a lot of interest for their hardware outside the US/Japan market. Same
> > goes for the rumoured 4
Hi lists,
I am trying to connect SIP Phone and H323 Phone. I can call to from
SIP-Phone to H323 with clear voice. But I can't hear the voice calling from
H323-phone to SIP-phone. The ring and hookup function is OK. I am using
chan_h323 driver. I also tried changing codecs, g711u and g723.1. The re
On Wed, Aug 06, 2003 at 09:59:18AM -0500, Martin Pycko wrote:
> You're looking for libncurses-dev and in libpri you can remove -Werror
> from libpri/Makefile or cvs update libpri (it should be fixed)
fixed, in that they removed -Werror, or fixed in that they updated the code
to eliminate the warni
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