I'm using a Sangoma A101 card alongside an older TDM400 and they seem to be
playing nice. I've had it in production for a few months now with no
problems.
Thanks,
Reid Forrest, CISSP
Max-IS Inc.
[EMAIL PROTECTED]
Direct/Cell: 321-214- Main: 407-786-9600
-Original Message-
From
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Sunday, September 18, 2005 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AstriCon 2006 Location
Senad J
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin Bockman
Sent: Saturday, September 17, 2005 2:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AstriCon 2006 Location
Matthew
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming
Sent: Tuesday, September 13, 2005 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM400P stops answering
Andy
:
Waiting for zap to come online...Error: missing /dev/zap!
Wha am I doing wrong?
[Reid Forrest]
Check out README.udev in the zaptel source directory. You need to make
modifications to the udev config files.
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Asterisk
i need to create a group extension, to make calls to 6 sw
phones, but i
need to know if asterisk can do help me to get a unique
number and check
what extension has received less calls than the others, and
pass the new
call. We got a call center and want to know if we can distribute
I'm running Asterisk 1.0.5 stable. Before that I have run
1.0.3, 1.0.1, 1.0 and many CVS versions all with the same
symptoms.
Thank you,
Reid Forrest, CISSP
Max-IS, Inc.
[EMAIL PROTECTED]
ofc: 407.786.9600 x1200
cell: 321.439.8903
From: [EMAIL PROTECTED]
[mailto:[EMAIL
This is not normal; I do *not* have this issue with NuFone
and I have placed a
ton of calls through them daily for the past year. I don't
recall having
this problem with voicepulse connect when I used them, nor do
I have the
issue with iax.cc for inbound calls.
I'm experiencing
Check the load on your server(s).
Load average is always at or near 0. This is on a dedicated machine doing
nothing but routing calls. No voicemail, music on hold, etc.
I noticed something in a packet capture that may or may not be significant.
When I place a call, the capture shows about
. Someone wrote in response to the last post saying that the audio
path probably wasn't set up yet. I think this is the symptom, but I'm
wondering what's the cause, and if there's a fix.
Surely I'm not the only one who's having a huge problem with this. Can anyone
help?
Thank you,
Reid Forrest, CISSP
provider? I'm using Asterisk 1.0.3 and
have tested on different systems, different providers, different phones, etc.
Thank you,
Reid Forrest, CISSP
Max-IS, Inc.
[EMAIL PROTECTED]
ofc: 407.786.9600 x1200 cell: 321.439.8903
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Samudra E. Haque
Sent: Sunday, December 19, 2004 12:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialplan selection
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers
[globals]
X1000=SIP/1000
X1001=ZAP/1001
X1002=IAX2/1002
X1003=SIP/1003
[outbound]
exten = _123,1,Dial(${X${EXTEN:4}},10)
Oops, that line should read:
exten = _123,1,Dial(${X${EXTEN:3}},10)
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[EMAIL
I would be grateful if anybody could tell me what I should
tell Verizon
in NJ so they would enable disconnect supervision for my lines.
Apparently remote hangup notification or disconnect
supervision or
calling party control is NOT the magic phrase for them. Although
disconnect
On the surface, that sounds like an * problem, not sprint.
What are you using to interface to sprint (analog, bri, T1), which
cards in your * box (and associated config files)?
A fairly standard telco operating approach is _not_ to
provide any answer
supervision, and * works just fine
It's an analog POTS line connected to a TDM400 interface in
the * box. Config
files are set for kewlstart.
I've never heard of this problem. Analog FXO ports on Asterisk are
considered answered when Asterisk finishes sending the DTMF.
Do you have callprogress=yes in
in your plan. Additional outbound calls cost $0.039 / minute
each.
Although I haven't used them myself, I understand that VoipJet allow multiple
outbound calls within the US for $0.013 / minute.
Thank you,
Reid Forrest, CISSP
Max-IS, Inc.
[EMAIL PROTECTED]
ofc: 407.786.9600 x1200 cell: 321.439.8903
dtmfmode=inband
secret=xxpasswordxx
insecure=very
Thank you,
Reid Forrest, CISSP
Max-IS, Inc.
[EMAIL PROTECTED]
ofc: 407.786.9600 x1200 cell: 321.439.8903
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