Have you done a ps -elf to see if the process has a parent that is
re-launching or preserving it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, November 19, 2008 1:58 PM
To: asterisk-users@lists.digium.com
Subject:
Nov 2008, Danny Nicholas wrote:
Have you done a ps -elf to see if the process has a parent that is
re-launching or preserving it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, November 19, 2008 1:58 PM
does not exist in /proc/modules
(sigh). In fact /proc/modules is empty.
[EMAIL PROTECTED] init.d]# ls -ltr /proc/modules
-r--r--r-- 1 root root 0 Nov 19 14:46 /proc/modules
j
On Wed, 19 Nov 2008, Danny Nicholas wrote:
Your could try this
History|grep modprobe
Rmmod XXX where xxx
In my meetme.c, users is defined as an int on line 328. This gives a
possibility of 35768 people in a conference. If you cbanged that to a
signed char, you would limit it to 127.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent:
Just set up a new spy in the dialplan that performs a Background on the
sound file, then hangs up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson
Sent: Thursday, November 20, 2008 4:34 PM
To: Asterisk User MailList
Subject: [asterisk-users]
Here is a Dirty solution - create a PERL or other script to listen for
changes to voicemail DB/Dir. When VM is deleted, launch script to turn off
Cisco MWI (should be simple since you are turning on with script). Not
Best solution, just workable one. I'm doing similar thing with my VM - I
look
Armed with a little more information, here is a more realistic reply.
In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the
max value in line 870 to 0x7fff.
Therefore changing line 870 would allow you to limit the maxusers.
-Original Message-
From: [EMAIL
You could trying changing this in sip.cfg
AES voice.aes.hs.enable=0
To
AES voice.aes.hs.enable=1
It's at line 324 in mine. Results not guaranteed.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 10:28 AM
To:
You could try un-commenting duplex=2 in rpt.conf and changing it to
duplex=3.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 11:05 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] upgrade
Just a guess, but since extensions.conf is basically a card file, there
may be a 80 character limit to the line or data size.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Loic Didelot
Sent: Friday, November 21, 2008 11:52 AM
To:
Do you use the Asterisk GUI? Changes from it can mess with contexts in the
dialplan (extensions.conf) and the hints need to remain in the [internal]
context.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, November 26, 2008 6:33 AM
To: 'Asterisk
Just speaking theoretically, you should be able to do a Zap/SIP bridge just
like using a TDM???. How does this show up in the CLI interface (core show
channels)?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Irfan Malik
Sent: Wednesday, November 26, 2008 8:11 AM
Assuming you have caller id, you can call MeetMe with different parameters.
You could also write an AGI to handle the announcements and leave meetme in
Silent (No Announce) mode.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giedrius Augys
Sent: Wednesday, November
What are the lines in your dialplan for using the Mobile line? For example
exten = NXX,1,Dial(Zap/g1/${EXTEN},60)
dials a local (7 digit) number using Zap Group 1, waiting 60 seconds for
connection.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
: +92 42 5785703-8 Ext: 196
Web: www.tcm.com.pk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 7:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mobile
, November 26, 2008 9:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] language and meetme issue
2008/11/26 Danny Nicholas [EMAIL PROTECTED]
Assuming you have caller id, you can call MeetMe with different parameters.
You could also write an AGI
The phone should renew itself to asterisk periodically even after a
reboot. My setup renews the connection every 2 minutes (non-critical,
small shop).
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Wednesday, November 26, 2008 9:24 AM
To:
-Commercial Discussion'
Subject: Re: [asterisk-users] Hints stopped working suddently
Not at all, I do everything with vi
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
Sent: Wednesday, November 26, 2008 8:51
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
appears when the
phone is disconnected, but the status inuse doesn't when on a call.
That unavailable works fine is some sort of proof that everything is
setup properly.
Mike
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Danny
Nicholas
*Sent:* Wednesday, November 26
The quick answer is that your realtime isn't transmitting full frames.
This message occurs when the number of bytes from the frame read isn't
divisible by 4. Changing the rtpchecksums in rtp.conf might correct this.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
- Non-Commercial Discussion
Subject: Re: [asterisk-users] RTCP too short
realtime? I'm using static config files, no realtime.
- Original Message -
From: Danny Nicholas mailto:[EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com
Non-Commercial
Iax2 provision would seem to be a harsh but simple way to do it. Iax2
provision and iax2 prune seem like kinder candidates, but I haven't gotten
into the iax2 branch of * yet.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Augustyn
Sent: Friday, November 28,
Message -
From: Danny Nicholas mailto:[EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com
Non-Commercial Discussion'
Sent: Friday, November 28, 2008 1:22 PM
Subject: Re: [asterisk-users] RTCP too short
Double check your config files. Rtp.c is a real
Why wouldn't this work?
exten = _911,1,Hangup(Zap/1)
exten = _911,2,Dial(Zap/1/ww911,60)
exten = _911,3,Hangup()
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Messina
Sent: Friday, November 28, 2008 3:48 PM
To: Asterisk Users Mailing List -
plan except implicitly.
Danny Nicholas wrote:
Why wouldn't this work?
exten = _911,1,Hangup(Zap/1)
exten = _911,2,Dial(Zap/1/ww911,60)
exten = _911,3,Hangup()
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Messina
Sent: Friday, November
You shouldn't open text your password. Shouldn't IP on Asterisk 2 be
1.2.3.4?
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shaun Wingrin
Sent: Monday, December 01, 2008 4:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Inbound calls from
Festival is a free voice that sounds like a machine. Cepstral is a fee
based human voice ($30 USD per voice per CPU). They are similar in that
they both produce mechanically timed output. IMO, you should use festival
if this isn't a customer based interface. If it is a CBI, use cepstral and
if
You can send an IM to the phone with a text message. Assuming that the
phone has more than 1 line and at least one is open, the call should go
through without effecting the existing call. To do this from the dialplan,
you could set up something like this:
Exten = 411,1,Dial(SIP/100,1)
Exten =
This seems to be an AGI/Music on Hold solution to me. For parking to work,
you would have to know which lot you parked the call in and pick it back up
when done, assuming that another user did not pick it up and that the caller
did not hang up.
From the dialplan, you would call an AGI. The AGI
The way I made this work was to set up 200 as my parker and I do transfer,
200, transfer.
exten = 200,1,Answer
exten = 200,n,Park(701)
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Wednesday, December 03, 2008 10:33 AM
To: 'Asterisk Users Mailing List -
Subject: Re: [asterisk-users] Call parking
On Wed, Dec 03, 2008 at 10:56:48AM -0600, Danny Nicholas wrote:
The way I made this work was to set up 200 as my parker and I do
transfer,
200, transfer.
exten = 200,1,Answer
exten = 200,n,Park(701)
That will work but only for one call park slot
Discussion
Subject: Re: [asterisk-users] Call parking
On Wed, Dec 03, 2008 at 11:13:49AM -0600, Danny Nicholas wrote:
This actually works for multiple slots. When 701 is occupied, * finds
next
defined slow.
Does it announce what that slot is before doing it?
Rob
--
Robert Lister - London
Hold down 2,4,6,8 and * at the same time. This is the 501 reset key
sequence.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, December 04, 2008 6:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Have you checked voip.org? They have this kind of information for a
Polycom, so they probably have similar information for the Cisco 79x1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lincoln
King-Cliby
Sent: Friday, December 05, 2008 11:10 AM
To:
Good programmers can diagram the most obfuscated code. It's part of the
job description.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: Friday, December 05, 2008 12:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Debug level 2 (core set verbose 2) works well for me.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, December 05, 2008 3:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] CLI and choice of
The 100,000,000 calls without a crash are more impressive to me than the
1000 days of uptime. Mine crashes on crazy things like dynamic conferences,
etc. :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
LaCoursiere
Sent: Monday, December 08, 2008
Looks like someone messed with a umask. I would guess that freepbx can't
read non-permissioned files. The umask for msg0005.txt was 0770. The
umask for msg0005.gsm was 0777. I'd check the shell commands that run in or
under both processes for changes. BTW, you should probably be running
This sounds like a job for a VB.NET programmer. The program would run like
a DDE server and ftp a call file to your asterisk server on the desired
action.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Fife
Sent: Monday, December 08, 2008 3:04 PM
To:
5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [asterisk-users] 'dialer' application to trigger call
betweenhardphone and number
There are a few web-based ones - is that an option at all?
PaulH
Danny Nicholas wrote:
This sounds like a job
Modprobe wctc400p will load the module. You will then need to (re)start
asteriskl.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 09, 2008 8:13 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] about trasncoders
know but i dont want to write modprobe every time i reboot the server...
there is a file but i cant remember the name...
2008/12/9 Danny Nicholas [EMAIL PROTECTED]
Modprobe wctc400p will load the module. You will then need to (re)start
asteriskl.
_
From: [EMAIL PROTECTED]
[mailto
know but i dont want to write modprobe every time i reboot the server...
there is a file but i cant remember the name...
2008/12/9 Danny Nicholas [EMAIL PROTECTED]
Modprobe wctc400p will load the module. You will then need to (re)start
asteriskl.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL
http://linux.die.net/man/3/strftime http://linux.die.net/man/3/strftime
has an explanation of this function.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Tuesday, December 09, 2008 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Core show channels will show you this. Here is an example
Channel Location State Application(Data)
Zap/1-1 [EMAIL PROTECTED]: Dialing AppDial((Outgoing Line))
SIP/104-085ff278 [EMAIL PROTECTED] RingDial(Zap/g1/ww2975000|60)
2 active channels
You've checked that another asterisk is running (ps -ef|grep asterisk)?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Berry
Sent: Wednesday, December 10, 2008 9:06 AM
To: Asterisk Users
Subject: [asterisk-users] a problem on Ubuntu with Asterisk
A bit of clarification on my previous answer that may help with some of
these replies:
Asterisk is theoretically launched at start up or some other time with
/usr/sbin/asterisk
-g causes core dump
-vvv causes verbosity level to be set
When you try to run your command, it should always include
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Gibbons
Sent: Wednesday, December 10, 2008 1:02 PM
To: 'Asterisk Users Mailing List -
Two things - Polycom phones require specific images sizes (480x180 pixels I
thinkg) and the sip.cfg has to allow presentation of images.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce
Sent: Wednesday, December 10, 2008 1:25 PM
To: Asterisk
Thanks to all of you toppers we can now plan on any message with top or
post being treated as spam. Some of us actually read these threads to
learn, not just to hear ourselves talk. If you really have to be top
somewhere, go to FoxSports.
-Original Message-
From: [EMAIL PROTECTED]
If callers need to just listen, you could run meetme with the -l mode.
Otherwise, you might try the -o mode (optimize, mute non-talker) or -m (set
initially muted).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessandro
Try this first:
Cat /var/log/asterisk/messages|grep channels unavailable
Once you get grep output from this, changing the grep on the tail command
should produce the desired results.
Since the tail -f is a dynamic situation, it is much easier to make the test
on the fixed cat command.
This seemed to be specific to 1.4.19 and prior.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled Chehab
Sent: Friday, December 12, 2008 6:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial
You should try these steps
1. core show application park from the CLI interface
2. look at features.conf
3. one of these should offer the hint you seek
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Did you make a [xyz] context in extensions.conf? if the sip.conf doesn't
find the content, it drops back to default.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Friday, December 12,
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, December 12, 2008 9:26
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Follow up on parking
You should try these steps
1. core show
Of Danny Nicholas
Sent: Friday, December 12, 2008 15:31
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Follow up on parking
After some research, it seems that asterisk builds a dynamic context called
[park-dial] and puts a callback for the parker into line 1
write
it...
David
2008/12/15 Danny Nicholas da...@debsinc.com
This appears to be the case. If someone else know how, please feel free to
share.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Monday, December 15
OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF
THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M
@#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT
MSN IS FOR.
-Original Message-
From:
This isn't what you're specifically looking for, but if you get an odbc
connection to the database, you can use that logic to do this. Try a google
on pgsql odbc connection asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Have you done a sip set debug, then sip reload? Do you have a range of
1-2 open in your firewall? Asterisk will poke out through 5060
but has to get a random response back in the 10-20K range (you can narrow
this)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
with sip debug info. I ran sip debug
again. The logged info is identical to what I was getting before I
changed the codec allow/disallow settings...
I think this is an Asterisk behind a NAT configuration problem, but if I
could be wrong.
Bud
On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote
What does your extensions.conf look like for this call? If you can insert a
ww into your Dial command (ie, change 18005551212 to ww18005551212) this may
improve your dialing behavior.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
address. Also, junctn.net's sip connection is working no
problem. I'm stumped. :(
Any ideas out there?
Thanks for the help so far...Still plugging away...
Bud
On Fri, 2008-12-19 at 17:00 -0600, Danny Nicholas wrote:
Check you codecprobe settings
http://lists.digium.com/pipermail/asterisk-dev
Not a solution, but a work-around. You could write a routine in Perl or C
or something to monitor the database and send out the voicemail via IMAP
when it comes in. I do a similar thing to increase the volume of a received
voicemail (WAV) and send it out to an Iphone.
-Original Message-
Hi Gang,
I'm trying to make an application to upload a tiff via a web
interface, slap a cover page onto it, merge the two into a new tiff and send
it out via txfax. I'm able to get it out to a fax machine using this
sequence:
/usr/bin/tiff2pdf /tmp/faxout/1229978819_filea.tif -o
This is how you would do it in 1.4
Exten = s,n,Set(GLOBAL(PERIODIC_ANNOUNCE)=/var/lib/asterisk/sounds/foo)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Broyles
Sent: Friday, January 02, 2009 7:22 AM
No. This would be an until changed modification (either via asterisk
restart or another similar command). If you took the GLOBAL off of the
set, it would probably be a 1 call deal.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Try it this way:
exten = s,1,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue2)
exten = s,n,Playback(${PERIODIC_ANNOUNCE})
exten = s,n,Queue(CSR)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Broyles
Sent:
Was this the real Allison or Cepstral-Allison-8khz?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Thursday, January 01, 2009 5:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Perhaps this thread will offer the solution you seek:
http://forums.digium.com/viewtopic.php?t=16253highlight=periodic
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Broyles
Sent: Friday, January 02,
Going on the assumption that CSR is in queues.conf and that
queue-periodic-announce is in /var/lib/asterisk/sounds, you could create a
CSR2 in queues.conf, a queue-periodic-announce2 in /v/l/a/s and make the
dialplan call CSR2 instead of CSR.
-Original Message-
From:
Perhaps there is something in queues.conf you can tweak. If you had one of
the ambitious asterisk setups where your configs were in a database instead
of a file (I'm not one of those BTW), you could use a DB command to set
this.
-Original Message-
From:
This might do the trick
exten = s,1,Answer()
exten = s,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue-announce)
exten = s,n,Set(PERIODIC_ANNOUNCE_FREQUENCY=60)
exten = s,n,Set(JOINEMPTY=1)
exten = s,n,Queue(CSR)
exten = 18001231234,1,Answer()
exten = 18001231234,n,Set(PERIODIC_ANNOUNCE=30)
exten =
Yes, but if you do, you will lose it in a future upgrade (if that matters to
you).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pedram M
Sent: Monday, January 05, 2009 4:41 PM
To: Asterisk Users Mailing List -
Why not just make a moh file of a ring-tone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mateusz
Pawlowski
Sent: Tuesday, January 06, 2009 1:59 PM
To: asterisk-users@lists.digium.com
Subject:
Which release of * are you trying to connect to? 1.6 has Cell capability
and the skinny option is available on 1.4 and 1.6. Just a thought.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Broyles
Sent:
You can install the custom mp3 player or just convert your mp3's to gsm with
lame. I did this with a free Beethoven overture download and it sounds
pretty good over the phone line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
You have to enable presence on the polycom phones, then they will read hints
from the default context of your dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Willis
Sent: Thursday, January 08,
RED is just off-hook or unavailable (at least in my shop).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of fidibus83
Sent: Monday, January 12, 2009 11:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] a
Provider 2 is dropping into a new context than Provider 1. The $EXTEN is
probably coming in from P1 as XX and P2 as AXX. Check your incoming
and default sections of extensions.conf.
_
From: asterisk-users-boun...@lists.digium.com
Why are you using a text message when you could be recording a message and
sending it out? This would possibly be clearer than a read-and-callback
scenario?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Since we are all learners here, you can download the Java stuff for free
from sun, but you'd need about as much time on the Java as you spend on *.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Why not use call-conferencing? If you transferred your call into a
conference room, you could join the conference from any extension on your *.
When the caller hangs up, just end the conference.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Here's a working scenario from my asterisk -
I have a static conference 6350 set up with no password. When a call comes
in, I transfer it to 6350. I can then access this call from any extension
by dialing 6350.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
What about Chanspy()?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Geoff Lane
Sent: Thursday, January 15, 2009 2:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, January 15, 2009 2:45 PM
If youre using the GUI it will hang it up. Otherwise sip reload might do
it.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy
Dobrovolskyy
Sent: Friday, January 16, 2009 12:00 PM
To: Asterisk Users Mailing List
Yes, BUT .. not 100% and discontinued in 1.4.22 on ...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, January 16, 2009 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial
Why not do a zap restart instead of restarting asterisk? You could write
an AGI to do the ZR when the condition occurred and lines where empty.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro
no message waiting ??
Sounds like an awful hack. What does DAHDI do that Zaptel does not?
Sounds more like a post for the bugs list
On Fri, Jan 16, 2009 at 4:48 PM, Danny Nicholas da...@debsinc.com wrote:
Why not do a zap restart instead of restarting asterisk? You could
write
an AGI to do
It seems to me that $FAXFILE lives in the 6403 context, not the n, so it
would still be useable. If this isn't the case, I'd call an AGI script in
the I context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Heres a link to a page that manually does this.
http://blog.asteriskguide.com/bandcalc/bandcalc.php
The general rule of thumb is that * uses 6-30Kb per SIP/IAX channel
depending on the codec. If you wanted a good live estimate, you could write
an AGI that uses Asterisk rx core show
Greetings all,
I'm trying to connect to an ATT teleconference, but the
call is never marked as ANSWERED by asterisk and therefore won't bridge and
continue. The only work-around I've come up with so far is to dial like
this:
Exten = 744,1,Dial(Zap/g1,,p)
The private mode
I'd try
xxx...@domain.com:Password:xxx...@domain.com
mailto:assword%3axx...@domain.com @domain.com
Or
'xx...@domain.com':Password: mailto:assword%3axx...@domain.com
'xx...@domain.com'@domain.com
_
From: asterisk-users-boun...@lists.digium.com
Depends on codecs and frames you've selected. Zap use T.30 and SIP uses
T.38.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
amir...@namche.com
Sent: Friday, January 23, 2009 12:36 PM
To:
-users] Zap connection problem
On Thu, Jan 22, 2009 at 08:28:18AM -0600, Danny Nicholas wrote:
Greetings all,
I'm trying to connect to an ATT teleconference, but
the
call is never marked as ANSWERED by asterisk and therefore won't bridge
and
continue. The only work-around
Does CENTOS have a rescue option?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, January 22, 2009 4:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Did you read the source for app_voicemail? Line 239 says you have to set
locale in the config and have the sound file einE. Of course an easier way
would be to locate the 19 day and month files and just replace them with
German equivalents (assuming that 26 and 2009 sound the same in a German
You've tried a sip reload from CLI and rebooted the phones?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Sent: Monday, January 26, 2009 8:51 AM
To: asterisk-users@lists.digium.com
Subject:
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