On Fri, 22 Aug 2008, larry wrote:
> HI
>
> Here is a question about the fxs of the zaptel card which is set a
> number to use in the inter as common analog phone. When I also use
> ${CALLERID(num)}to get it's number, it also could not be done. At this
> time ,the fxs phone does not get any re
On Wed, 14 May 2008, Roberto Milani wrote:
> From: OCG Technical Support <[EMAIL PROTECTED]>
>
> > Permissions? Try running msmtp from the asterisk account? (Assuming
> > that is how you have it setup)
> > I don't know msmtp - but is there a maillog equivalent?
> >
> > MD
>
> thanks for the re
On Mon, 14 Apr 2008, Jeremy Malcolm wrote:
>
> I am having trouble with chan_zap.so not loading. When I load it from
> modules.conf, Asterisk bails out without any error message. When I
> load it from the console, it just says "Unable to load module
> chan_zap.so" no matter what verbose level I
On Mon, 17 Mar 2008, Steve Totaro wrote:
> To get back on track, this is a little more complicated to get working
> on other distros than I previously thought.
>
> I am able to get the GUI to a page that says "Upgrade Druid" but it
> always fails with "Upgrade database failed for version 1.0.0-1.
On Mon, 25 Feb 2008, Tilghman Lesher wrote:
> On Monday 25 February 2008 10:04, Vincent wrote:
>> On Fri, 22 Feb 2008 09:15:35 -0600, Tilghman Lesher
>>
>> <[EMAIL PROTECTED]> wrote:
>>> Generally, the rule is that you can't remove any of the res_*
>>> modules.
>>
>> Thanks for the tip. At this p
Sure that isn't supposed to be 'zap show status' or 'zap show channels'
I don't have a Sangoma card but...
Brett
On Thu, 3 Jan 2008, [UTF-8] Ra??l G??mez C. wrote:
Hi list,
I've just compiled and installed Asterisk 1.4.16 and when I try to run "zap
show" I get the message "*No such command
On Thu, 22 Nov 2007, Edwin Kariuki wrote:
> Hi,
>
> I have a voip platform that has a SIP server where about 450 sipura
> phones & adaptors register. On two occassions some phones (which were
> previously working) have refused to register with certain IPs but when I
> change the IP the phones r
On Sun, 18 Nov 2007, broadband Voice wrote:
> I looked through /etc/asterisk and could not find the folder sampl.call.
> On 11/18/07, Tilghman Lesher <[EMAIL PROTECTED]> wrote:
>> On Sunday 18 November 2007 10:20:18 broadband Voice wrote:
>>> I have created a conference call solution for a client
On Wed, 14 Nov 2007, Matt wrote:
> I have asterisk 1.2.18 running on a new system we just installed.
> Although I've used AGIs many times in the past, I'm stumped on this one.
> All seems fine. If I run the script from the command line it works as
> expected:
> However, when actually running in p
On Thu, 25 Oct 2007, Frederico Madeira wrote:
> Philipp
>
> This didn't wotk.
>
> Let's suppose that my sip extension 3000 want to call to (302).123.3211
> I need a rule in extensions.conf to match with this number, right ?
> So, I can't use rules that you advice.
>
> My problem is only for outbou
Pablo - You said you have 1/2 E1 - which half???
That might be your problem. Unless 1/2 E1 means something else...
Asterisk normally dials out on the low end unless you specify
G instead of g ??? or something like that.
Brett
___
--Bandwidth and Colo
On Saturday 16 June 2007 02:24 am, Senad Jordanovic wrote:
> Paul Hales wrote:
> > GUI bad! CLI good!
> >
> > PaulH
>
> Really...?
>
> So explain why every major PBX manufacturer has GUI of some sort?
> Surely they would have had CLI only if GUI is "bad"!!!
>
>
> Senad
Senad - it is really to cove
I have been getting these for awhile now in my log files.
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk na
On Sunday 26 November 2006 12:04 pm, Androtech wrote:
> where is the "calling" button?
> I do not have it. I just can call typing numbers. The manual says that I
> should press "#" at the end of the number, but it does not work
>
> > On Saturday 25 November 2006 09:38 am, Androtech wrote:
> >> Hi a
On Saturday 25 November 2006 09:38 am, Androtech wrote:
> Hi all,
>
> I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028.
>
> This IP phone is registered in an Asterisk PBX and I've a problem when I
> dialing internal number. If I dial an internal number, like for example
> 102, the
On Wednesday 22 November 2006 08:43 am, Michael Welter wrote:
> Is there a programmatic to to trim the silence from the beginning and
> end of a recording? From a .wav file? From a .ulaw file?
>
> Thanks,
try "man sox" - look for 'silence'
Brett
___
-
16 matches
Mail list logo
