Thank you very much.
Dan
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, July 26, 2003 6:48 AM
Subject: Re: [Asterisk-Users] time and date stamp in voicemail
> On Friday 25 July 2003 14:12, Andy Hester wrote:
> > Dan,
> > the page
Hi,
Check if mpg123 executable is copied in the directory /usr/bin
It is not there by default.
BR,
Dan
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, July 26, 2003 2:23 AM
Subject: [Asterisk-Users] can't get musiconhold to work
> I can't seem to
IIRC I had the same problem becasue the package will install the mpg123 binary to
/usr/local/bin and * seems to look in /usr/bin so just copy the mpg123 executable to
/usr/bin and it should work..
Later..
> I can't seem to get musiconhold to work. I'm running asterisk on a RH9
> box, I have t
Hi,
I am looking at using a Netjet-S ISDN card with Asterisk, and would like to
know if it is possible
for asterisk to determine the dialled number (From the Indial Number range)
and route the call accordingly.
Would I just set up an extension number based on the Indial number?
Thanks
Jay
__
On Sat, 26 Jul 2003 17:13:42 +1000, Jay Tyndall wrote:
>
>
>Hi,
>
>I am looking at using a Netjet-S ISDN card with Asterisk, and would like to
>know if it is possible
>for asterisk to determine the dialled number (From the Indial Number range)
>and route the call accordingly.
YES
>Would I just
Actually one ofthese days I must actually look at trying the capi
drivers (just lack of time)...
Q for the list
Can you run more than one card with the capi drivers, I seem to
remember sighting an issue on this ??
Gary
On Sat, 26 Jul 2003 17:13:42 +1000, Jay Tyndall wrote:
>
>
>Hi,
>
>I a
I had the same kind of problem until I upgraded my asterisk-0.4.0 to
latest CVS. Then X-Lite kind of worked : I could hear and the
announcements and let a message in my voicemail.
But DTMF doesn't seem to work : I tried to log in my voicemail to hear
my messages, "dialed" the good "password", b
Perhaps it'd be good to have asterisk queue the email instead of sitting
there waiting for it to be sent?
roy
On Fri, 2003-07-25 at 17:27, Wade Weppler wrote:
> What Steve says... Also, check your hosts file for strange entries. This
> was the problem in our case. We had exactly the same sympt
Hi,
With Asterisk and SIP phones (Cisco ATA186, Grandstream BT102), I'm
having an issue with DTMF passing correctly to IVR systems like
customer support phone numbers, voicemail, etc.:
1) If I set DTMF to SIP INFO, DTMF works for ISDN4LINUX calls to IVR
systems with the CiscoATA186, but not wi
Hey AJ,
On Fri, Jul 25, 2003 at 19:23:50 -0400, [EMAIL PROTECTED] wrote:
> I can't seem to get musiconhold to work. I'm running asterisk on a RH9
> box, I have the mpg123 package installed. In my zapata.conf file I have
> the line MusicOnHold=default . In my musiconhold.conf file, in the
>
Scrive Paul Cheng <[EMAIL PROTECTED]>:
> What baffles me is why SIP INFO doesn't work with Asterisk AppVoicemail.
>
I'm using DTMF = info with my budgetones and they works great,
voicemail is ok, dtmf on remote ivr are ok too.
what * version are you using?
--
Matteo Brancaleoni
Espia Syste
Hey all,
I'm experiencing a problem with chan_sip on a multi-homed machine.
The machine has 1 interface to the rest of the world and 1 interface
on a local network. The local network has public IP-addresses, though,
and the IP-addresses of both interfaces are reachable from the outside
world, but
Wipeout
I'm using the exact mpg123 binary that you sent me. When I execute a
"whereis mpg123" it returns /usr/bin. To take it a step further I've done
"whereis mpg321" and "rpm -q mpg321" just to make sure mpg321 is not on
the system. The one thing that's confusing the heck out of me is the f
Hi,
Can someone confirm the format of the Dial string for a H.323 gateway using
chan_oh323? The format I have working is:
exten => 5000,1,Dial(OH323/h323:[EMAIL PROTECTED])
I have 5000 as a speed dial - the extension functions, but the voice
latency within the call to the analog phone (9945
On Fri, 25 Jul 2003, Yifang Dai wrote:
[...]
> I've successfully flashed 2 7940 with sip image, they can now talk to
> asterisk pbx, call each other, vm etc.
Cool.
> Now I'd like to get them talking to CCM via asterisk through the oh323
> channel.
>
> extensions.conf
>
> exten => 6107,1,Wait,2
>
Sorry I though you had compiled from source...
When * is running do "ps-aux | grep mpg123" and make sure it is actually running..
Later..
> Wipeout
> I'm using the exact mpg123 binary that you sent me. When I execute a
> "whereis mpg123" it returns /usr/bin. To take it a step further I've don
No instances of it running when I look at processes.
AJ
On Sat, 26 Jul 2003, WipeOut . wrote:
> Sorry I though you had compiled from source...
>
> When * is running do "ps-aux | grep mpg123" and make sure it is actually running..
>
> Later..
>
> > Wipeout
> > I'm using the exact mpg123 bina
Hi,
after cvs upgrading my * installation yesterday, the prompts in both
VoiceMail2 and VoiceMailMain2 have become silent. All I get is the initial
"blip" followed by the Voice taking breath and being cut off before she
has a chance to say "Comedian Mail".
All other prompts (ie the Playback appli
I'm using CVS from yesterday.
I just updated to today's CVS and no change. The Grandstream works fine
with SIP Info with everything but IVR calls through I4L. It was when
the Cisco ATA186 was set to dtmf=info that voicemail wasn't detecting
the DTMF--but i4l calls would work with the Cisco.
Is
I was finally playing around with the various databases,
blacklist and cidname. The format
for these are family name, key and value. Would that mean that the key
is the name to enter, and the value the number? For instance:
database put
blacklist “Kim Callis” 3235551212? And if that is in
Only things I can suggest is..
1. Execute it from a command line and make sure it runs.. If not you may hevr to
compile it from source..
2. Make sure you have a new line at the end of your .conf file cos * often freaks out
about that..
Other than that I don't know why its not working for you..
I noticed in the source for app_meetme.c,
that there is an arg to allow someone to be an admin.
What exactly does that mean? What extra privileges do they gain? And
furthermore, how is one made an admin in a room?
3) someone needs to hack the playing of precompressed .gsm files for
MusicOnHold. This would save a tremendous amount of resources just to
play frigging music.
Jeremy McNamara
WipeOut . wrote:
Only things I can suggest is..
1. Execute it from a command line and make sure it runs.. If not yo
-- Called ttyI0:09854433
-- Modem[i4l]/ttyI0 is busy
== Everyone is busy at this time
Everything works fine in minicom - i can call.
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Is there something I need to do to make asterisk start?
On Sat, 26 Jul 2003, WipeOut . wrote:
> Only things I can suggest is..
>
> 1. Execute it from a command line and make sure it runs.. If not you may hevr to
> compile it from source..
>
> 2. Make sure you have a new line at the end of your
hi ..
i've just converted myself back to a newbie by trying to experiment with
some new stuff .. I have connected two grandstream Budgettone 100 phones
to my asterisk, and trying to experiment with them ..
I am trying to get into the asterisk sample basically ..
when I dial 1000 asterisk receiv
Hi,
I have asterisk-0.4.0 running. When I make a
call between an ATA186 and Asterisk using ulaw or alaw codec, all is
fine.
I installed the CVS version and tried the same
thing but the voice is choppy. The installation was done on the same linux
server. The stats on the ATA186 show no
My sip .conf entry for the gs phone looks like this..
[2010]
type = friend
context = local
sectet = xxx
host = dynamic
callerid = Name <2010>
Your sound issue sounds like a codec problem, make sure you have not disabled ulaw or
alaw cos these are the only codecs that work with the GS phones.. Un
Hello-
I've been writing a number of AGI scripts in Perl, and so far everything's
working ok.
However, yesterday I tried the AGI command "RECORD FILE" for the first time,
and my channel locked up. Trying to stop asterisk produced a segmentation
fault. There may be a bug here, but first let me m
ANNOUNCEMENT: Bug Tracker/Feature Request System
http://bugs.digium.com/
Digium has introduced a bug tracking and feature request system for
Asterisk developers and users. Due to the increased traffic on the
mailing list, and an inadequate number of hours in the day to parse
it, it has been deci
I've also played around with the language support but I never got it to
work either.
-bill
- Original Message -
From: "Panagidou Anna" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, July 25, 2003 3:15 AM
Subject: [Asterisk-Users] SetLanguage application doesn;t seem to work i
My compile of asterisk is bombing out in /usr/src/asterisk/res . I'm
running debian with kernel 2.4.21 (custom). Zaptel and libpri appear to
make okay.
Any suggestions?
When I reboot I don't see any indication that the zaptel modules are
loading. I don't know if the final bits of a successful
At 10:31 AM 7/26/2003 +0200, you wrote:
I had the same kind of problem until I upgraded my asterisk-0.4.0 to
latest CVS. Then X-Lite kind of worked : I could hear and the
announcements and let a message in my voicemail.
But DTMF doesn't seem to work : I tried to log in my voicemail to hear my
m
can you send the output from the compiler? without it, it might be hard
to troubleshoot
On Sun, 2003-07-27 at 01:37, Tim Petlock wrote:
> My compile of asterisk is bombing out in /usr/src/asterisk/res . I'm
> running debian with kernel 2.4.21 (custom). Zaptel and libpri appear to
> make okay.
>
Tilghman,
I applied your voicemail_prompts patch and it works like a charm. Thanks
for donating the code and thanks to those that donated the voice prompts!
Another win for Asterisk
Sincerely,
Andy Hester
Consero
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAI
I've merged some changes from Michael Manousos that should improve sound
quality on non-zap channels, including music on hold. I'd like to hear
back on or off list about your experiences with the new code. Thanks!
Mark
___
Asterisk-Users mailing list
Hi,
I put a TE410P card in a machine (a Tyan 2665 with 2x2.4GHz Xeons). A
red flashing light circles around the 4 RJ48C sockets. I load the
wct4xxp driver, and the flashing light stops. Whether I connect an E1
signal or not, no lights are shown, and no alarms are reports in the
/proc/zaptel/XX
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