Hi,
I am using Trxibox 2.6 latest ISO install.
Following is the output of : sip show channels
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
Hold Last Message
212.53.40.40 0218245 6cfb845d050 09011/0 0x0
do you have that user 1006 defined by IP ?
does it have mailbox= also defined ?
my wild guess is that there's unchecked voicemail and asterisk tries
to initialize sending NOTIFY MWI messages
you can't remove these messages they remove themselves after some timeout
Martin
On Sun, Sep 27, 2009
do you have that user 1006 defined by IP ?
*I have a user 1006.
Its not defined by IP.
*
does it have mailbox= also defined ?
*Yes. 1006 has a Mail box*.
my wild guess is that there's unchecked voicemail and asterisk tries
to initialize sending NOTIFY MWI messages
*I will delete all
my wild guess is that there's unchecked voicemail and asterisk tries
to initialize sending NOTIFY MWI messages
*I will delete all messages from the Mailbox and see if 1006 is removed
from the listing.*
Just checked, no messages in 1006.
Any other reasons!
Thx
Sanjay
I have a SPA742, which can autoanswer a call
In the dialplan, I have this:
exten = 28,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = 28,2,dial(SIP/36)
Now I want some external event initiate a call to that phone and play a
message.
I have been thinking of dialfiles, but I believe there is
With best regards
Abdul Ahad Anwer Khan, M.Sc(CME, in progress)
University of Applied Sciences Offenburg Germany
Phone:+497814748226
Mobile:+4917623468462
From: abdulahadan...@hotmail.com
To: asterisk-users-boun...@lists.digium.com
Subject: New in asterisk
Date: Sun, 27 Sep 2009 14:50:59
Martin wrote:
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
But I generate another unique file in [fax-tx] just before I try to send
the confirm.
Here's the first call:
-- Executing
On Sat, 2009-09-26 at 22:47 -0700, Dave Platt wrote:
Isn't an SSL based tunnel all TCP?
There seems to be a good deal of feeling (and evidence) that
trying to use TCP as the container for a tunnel is likely
to cause more trouble than it solves. Yes, the TCP layer
will make the tunnel
2009/9/27 Leif Neland le...@neland.dk:
Can I, via a callfile, or command-line parameters to Asterisk start a
dialplan-script?
eg asterisk -someflag execute callalert
then in dialplan
[callalert]
exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = s,2,dial(SIP/36)
exten =
On Sun, Sep 27, 2009 at 12:07 AM, Gabriel Ortiz Lour
ortiz.ad...@gmail.com wrote:
Hi,
Is there a way to know for how long an agent is talking on the queue call?
(without keeping a timer myself... just asking asterisk)
Identify the channel at the CLI and then get its details via
core
I've been willing to give such a solution a try but the lack of time has
prevented it to date...
Are you using realtime for your SIP peers/users ? Would the failover
behaviour improve under such scenario ? (just a thought)
--
exvito
___
--
I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but SIP only. I
then downloaded and installed latest Zaptel and could not get Zaptel working.
So I downloaded Asterisk again and re-installed.
But still problems:
Here is my ztcfg output:
asterisk:/etc/asterisk # ztcfg -v
Ex Vito skrev:
2009/9/27 Leif Neland le...@neland.dk:
Can I, via a callfile, or command-line parameters to Asterisk start a
dialplan-script?
eg asterisk -someflag execute callalert
then in dialplan
[callalert]
exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten =
On Sun, Sep 27, 2009 at 05:17:33PM +0100, Angus Asterisk wrote:
I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but
SIP only. I then downloaded and installed latest Zaptel
I'll note that for many practical purposes, latest zaptel is DAHDI.
and could not
get Zaptel working.
At 09:17 AM 9/27/2009, you wrote:
What should I be looking at? Works ok for SIP but I want to get the
analog card working. It is a TDM04B.
Have you tried running genzaptelconf or whatever it's called?
Ira
___
-- Bandwidth and Colocation Provided
On 1.6.0.16-rc1:
I'm using app_fax.so to send a fax, and then send a confirm.
'send' = 1.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
2. System(env echo -e
Channel:DAHDI/g0/\\nContext:fax-tx\\nExtension: s\\nPriority:
1\\n
Hi All.
I have many days reading and research about asterisk and vicidial. I thing
this issue is about asterisk and doesnt about vicidial. Isn't it?
I have a problem with theses application (I already ask for help in vicidial
forums), but I can not fix it.
I have debian 5 with asterisk 1.2.24
I'm using QSIG between an NEC SV8300 and Asterisk (after giving up with
CCIS). Things work pretty well with the exception of issues on stations
on the SV8300.
When I call from Asterisk to a SV8300 station and I send my extension
as the caller ID number, it shows up on the SV8300 as OPERATOR.
I am unable to dial out over a Wildcard TDM400P. This was working previously,
so must have
messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
2.5.2.2.
When I dial, I see:
-- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1,
DAHDI/g0/9239220,300,) in
new stack
Andy Howell wrote:
I am unable to dial out over a Wildcard TDM400P. This was working previously,
so must have
messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX
2.5.2.2.
When I dial, I see:
-- Executing [...@macro-dialout-trunk:19] Dial(DAHDI/1-1,
I've got hints setup for my MeetMe conferences like so:
exten = _60X,hint,MeetMe:${EXTEN}
and they show up in core show hints like so
6...@dialtone: MeetMe:600State:Unavailable
Watchers 1
_...@dialtone: MeetMe:${EXTEN} State:Unavailable
Hi!
I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound.
One specialist on the forums asked me if I have DAHDI configured, he assumed
that this could be cause of choppy sound problem.
dahdi_test
Unable to open dahdi interface: No such file or directory
Do I need to
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