I had this issue- it's security on the files. I put a cron job that do
/bin/chmod 777 /var/spool/asterisk/voicemail/default -R
evey 1 minute, but there may be a cleaner solution.
Assaf Benharoosh
MCP, MCSA, MCSE
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[m
Bill,
Are
you sure there's an AGI enviroment variable that gives me that? I couldn't find
any:
-- accountcode = -- callerid = "Assaf Benharoosh"
<21> -- channel = SIP/26-f39a -- context =
extensions -- dnid = 45 -- enhanced = 0.0 --
extension = 45 -- lan
SIP/21-6735 part?
Thanks.
Assaf Benharoosh
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Hello,
Does anyone have
experience in HDLC setup with T100P ? I've looked around, including wiki pages,
and still having problem in the ztcfg stage (or even in the make
stage).
Do I need to do
something in the kernel for the HDLC first?
I'm using 2.4.20
kernel.
Thank
Thank you for making this clear for me.
Is there any solution for the mentioned phones?
Assaf Benharoosh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Saturday, November 13, 2004 1:41 AM
To: Asterisk Users Mailing List - Non
Does anyone else
experiencing this issue with those phones and have a
workaround?
Assaf Benharoosh
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Has anyone
experienced this problem? The C flag in Dial app doesn't work. I'm getting the
CDR record although it has C (reset CDR for this call). The C is even recorded
in the CDR record lastdata field.
Assaf
BenharooshMCP, MCSA, MCSE[EMAIL PROTECTED]
__
I made it work:
Sip.conf:
register => 1:@atlas-east.vonage.net:5060
[vonage]
type=friend
username=1
secret=
host=atlas-east.vonage.net
port=5060
allow=all
maxexpirey=15
dtmfmode=inband
fromuser=1
fromdomain=atlas-east.vonage.net
canreinvite=no
nat=yes
context=
Extensions.conf:
exten => _1,1,
Hi
all,
Does anyone know if
it's possible to continue talking to one member of an initiated conference call
on Cisco 79xx ? In other words- disconnect one of the
parties.
Thanks.
Assaf
rname Refresh State
sphone.vopr.vonage.net:5061 16464855183 15 Registered
Help anyone?
Assaf Benharoosh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 6:33 PM
To: [EMAIL PROTECTED]
Hi,
I actually gave up on the LineJack. I'm using Digium 4 FXO card- which
does the job pretty well.
Assaf Benharoosh
MCP, MCSA, MCSE
[EMAIL PROTECTED]
Frantic, LLC.
246 West 38th Street
2nd Floor
New York, NY 10018
T: (212) 302-5790
F: (646) 201-9418
C: (516) 805-7981
-Original Me
I've been getting the same type of answers for the past month.
Assaf Benharoosh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Wednesday, June 16, 2004 12:08 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voicepulse
They had about 5 down times today- between 2 hours and 20 minutes each
:-(
Assaf Benharoosh
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David H
Hickman
Sent: Thursday, May 20, 2004 3:13 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users
the bad
sound effect occurs in VoicePulse. (just spend lots of time finding that
out)
>Assaf Benharoosh>MCP, MCSA,
MCSE>[EMAIL PROTECTED]>Frantic,
LLC.
To whom it may
concern,
When dialing out an
800 number (888,866,877) through VoicePulse IAX you'll get a choppy sound. This
is not due to a problem on your Asterisk or your line- the bad sound effect
occurs in VoicePulse. (just spend lots of time finding that
out)
Assaf
BenharooshMCP, M
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