Hi all,
Quick question. Is there a way to have multiple people have an
extension, say 900, to their polycom 501 SIP phones on one of the blue
buttons to where when a call comes in, I can have it simul-ring and
folks can pick up the line on their phone? I'd like to set up a tech
support extens
Rob,
Interestingly enough, I'm using that same sample macro, and that line is
indeed in there, yet when I hit *, I hear the tone to leave a message.
Any ideas?
Phil
Phil,
Add this to your extensions (I have mine in a macro)
exten => a,1,VoicemailMain(${ARG1})
Hi all,
I'm sure this is a stupid question, but is there a way to check your
voicemail by calling your extension from the outside? When I call my
own extension from outside and hit pound or star, it just stops my
greeting and gives me the "beep". I'd like to call my extension and
press a key
I used _XX. Since it was used in the examples I got from voicepulse.
Maybe I can modify it so it's standardized by using 's'. Any idea why
they'd use something like that for incoming calls? Are you sure 600
would match _XX.? I thought _XX. Was just two digits.
Thanks for t
Hi all,
I'm trying to incorporate using the i extension in my callplan to
determine if someone enters an invalid extension. My internal
extensions are all 3 digits (100-104). The problem is, the callplan
doesn't see that say, extension 600 is invalid, it just goes back to the
beginning of the
Greetings,
Currently my asterisk box is using Voicepulse. It works fine with the
exception that people need to enter the 1+area code for local calls.
I'd like to get around this if possible. The following is what I have
in my extensions.conf..
exten => _1NXXNXX,1,Set(CALLERID(num)=616
I'm totally at a loss here. I can't get music on hold when placing
someone on hold or when dialing an internal extension. When I dial an
internal extension I hear ringing yet on my phone it shows little
musical notes like it thinks it's hearing music. What to do! :-)
Phil
Lee Jenkins wro
I've gotten this Polycom 501 pretty much licked, but I need to know if
there's a way in a dialplan to say if someone dials their own extension
it goes straight to voicemail and asks them for their password. I
thought I saw an example of this on the web but I can't seem to find it.
Any advice appre
as to WHY it stops suddenly! This is driving me nuts.
Phil
Phil, did you add letter 'm' to your dial options??
exten => _XXX,1,Dial(SIP/XXX,60,m)
Regards
Arlen Nascimento
On 12/20/06, Phil Finkler http://lists.digium.com/mailman/listinfo/asterisk-users> > w
Hi all,
Can someone point me in the right direction here. What I'd like to do
with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom
phones and after the 3rd digit is entered, it dials that extension and
b) dial 9 to get out like older PBX systems. Since my internal
extensions
I installed the asterisk-addons from source and installed them. It
looks like it copied format_mp3.so but I'm not sure if 1.2.14 addons are
compatible with asterisk 1.2.10. Also I unpacked the asterisk source
for the 3 MOH .mp3's and copied them to the appropriate location. Still
MOH is not work
Hi all,
I've got Asterisk 1.2.10 up and running on Debian using the back ports.
I noticed that it didn't come with mpg123 or depend on it and I believe
I read somewhere that asterisk now handles it's own mp3 playback? Is
this true? If so I must have a problem, because I hear no music when
put
Hey all,
I've been doing a lot of playing, and a lot of reading, and it seems
people are split as to whereas if they're running their favorite Linux
distro and asterisk or Trixbox. I'm getting closer to really looking at
a production environment and I'm just looking for any opinions. I'm
real
then apt-get -t sarge-backports install asterisk
(you can also pin-priority asterisk's packages, look at APT
documentation).
-Alex
On 12/10/06, Phil Finkler <[EMAIL PROTECTED]> wrote:
Hi all,
I've gotten asterisk installed on Debian only to realize that the
packaged version
: Re: [asterisk-users] Unable to open pseudo channel for
timing...Sound may be choppy.
On Mon, Dec 11, 2006 at 04:18:47PM -0500, Phil Finkler wrote:
> Any idea what causes the warning "Unable to open pseudo channel for
> timing... Sound may be choppy."? Any ideas what I need to re
Any idea what causes the warning "Unable to open pseudo channel for
timing... Sound may be choppy."? Any ideas what I need to resolve
this? I do have the zaptel module installed but don't have a zaptel
card. I'm guessing this has to do with ztdummy? I'm running Debian and
installed asterisk, z
Phil
On Sun, Dec 10, 2006 at 11:12:32AM -0500, Phil Finkler wrote:
> I'm running Debian 3.1 with the 2.6 kernel. I've also got the kernel
> source installed and compiled from source so I'm assuming I don't need
> the kernel headers. Can you elaborate on what "m
Hi all,
I'm pretty new to linux and compiling modules, but I've scoured the web
for help on compiling the zaptel modules from source and I get the
following error...
make -C SUBDIRS=/usr/src/modules/zaptel modules
make: *** SUBDIRS=/usr/src/modules/zaptel: No such file or directory.
Stop.
Does there seem to be a popular Linux distro folks use specifically for
Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with
Linux distros. In particular, I'm looking for a free, stable, well
supported distro that has a friendly community. Any advice appreciated.
Sorry for ask
I'm new to Linux, as I've been using Asterisk on FreeBSD via the ports
collection. My question is simple - for using the release branch of
Asterisk (1.2.13 for now), should I get in the habit of using svn to
retrieve the source or should I just download the tarball? Is there a
"best practice" or
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