Yup, one way to guarantee the code in the Asterisk base is to spend a
few months chasing the bug tracker.
I think this feature was simply implemented without taking the users
contexts in mind.
That's my judgment though but I agree with you it's wrong.
Martin
On Sat, Apr 4, 2009 at 11:26 AM
is to add features to the code without updating the documentation.
That is left to a random effort.
Martin
On Sat, Apr 4, 2009 at 7:31 AM, Roland Roland r_o_l_a_...@hotmail.com wrote:
Hi all,
a few month ago I got the task of setting up asterisk for my company.
I had 94 employee to set this up
I can make a similar logo (yet different) with the same words and
I'll be fine.
I can also use these words together and you can't do anything about it.
How about I challenge you - try to trademark just those two words.
It used to be ONLY $400 LOL
Martin
On Sat, Apr 4, 2009 at 11:52 AM, Ira i
I believe both ways can be used. If I'm to leave one/two sentence note
like I'm doing right now then top posting is fine.
Otherwise it's good to throw some context so the reader can understand
what is the answer for.
And of course trimming the ads and useless signatures is good too
Martin
with one FXO
module then have the user buy a $5 winmodem (yes, these new modems are
still ~$5). So I don't see this effort making much progress anyways.
Martin
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asterisk-users mailing
before.
Martin
On Thu, Apr 2, 2009 at 11:07 AM, Khaled W. Chehab kche...@xplorium.com wrote:
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
Hi Khaled,
app Dial clearly is coded to ignore the 180 Ringing being passed if
you have 'm' option to Dial and you do.
Try to take the 'm' out and see if 180 Ringing is passed to the A-leg.
So if you want MOH and then when 180 Ringing comes to turn it off =
you need a patch.
Martin
2009/4/4
in iax.conf with no password to access
the unsecured context.
Martin
On Sat, Apr 4, 2009 at 3:42 PM, Todd Reese trees...@gmail.com wrote:
Hi All,
Coming in to day, the logs on the asterisk server showed several entries
such as:
[Apr 4 15:25:16] NOTICE[9280]: chan_sip.c:14627
Well maybe turn the dahdichanname=no to yes...
And check if you can open cat /dev/dahdi/pseudo ... or better yet
maybe you're running asterisk with user asterisk
and it doesn't have access to /dev/dahdi/pseudo ... ? Meetme tries to
open that for timing source.
Martin
On Fri, Apr 3, 2009 at 10:24
He's already using domain feature but its logic is to override the
user's context even if it was predefined in sip.conf
Martin
On Fri, Apr 3, 2009 at 3:14 AM, Olle E. Johansson o...@edvina.net wrote:
Or you could use the domain feature, where you set a default context
per domain
Hi did list his lsmod and it doesn't show dahdi modules ...
For me it seems to be that dahdichanname=no ...
Martin
On Fri, Apr 3, 2009 at 4:17 PM, Carlos Chavez cur...@telecomabmex.com wrote:
Last time I upgraded Zaptel to DAHDI I had a similar problem until I
erased the zaptel modules
I meant zaptel modules ... no zaptel modules loaded on his system
Martin
On Fri, Apr 3, 2009 at 5:28 PM, Martin asteriskl...@callthem.info wrote:
Hi did list his lsmod and it doesn't show dahdi modules ...
For me it seems to be that dahdichanname=no ...
Martin
On Fri, Apr 3, 2009 at 4:17
Turn off callprogres=yes or have it configured properly.
It should fix your problem.
regards
Martin
On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas da...@debsinc.com wrote:
Greetings listers.
I’m running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones. My
is not matched against username, but against
ip/port.
It's matched against both IHMO in some order of priority.
Martin
I need to have multiple queues a user can be logged in, therefore I need
to limit calls to phones (otherwise an agent would get multiple calls
at the same time). Because
You're trying to register with the service but Asterisk is using the
default expiry value of 120 seconds (1.6.x version)
And your provider wants you to use minimum of 3600 seconds (1 hr)
add defaultexpiry=3600 to [general] section of sip.conf
That should help register...
Martin
On Fri, Apr 3
believe that agrees with Q921/Q931 specs.
Martin
On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote:
Hi everione,
I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
timer
)
Martin
On Fri, Apr 3, 2009 at 12:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Thanks Tzafrir;
But did not get where to find drivers? I have zaptel.
Hi All;
If I need to set the opermode to King Saudi Arabia,
what the name I
have to use? For example, to set it for kuwait then I
use
Because you're thinking as a tech geek and not as a businessman.
They want to build company awarness and sell the complete package and
that's why they have their own branded ISO.
Martin
On Fri, Apr 3, 2009 at 5:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
I actually wondered why
but not the other way
around.
Martin
On Thu, Apr 2, 2009 at 11:44 AM, Jerry Geis ge...@pagestation.com wrote:
Is there anyway a T1/PRI can ignore the ANSWERED signal and just go
straight from a dial command
to the call was answered?
I have a PBX that when calling a certain analog trunk it is not giving
me
check if you loaded the module
show modules like codec_g729
or simply try to unload/load codec_g729.so
Martin
On Thu, Apr 2, 2009 at 1:25 PM, criptos crip...@aullox.com wrote:
Humm... should the list would be magic again?
I have just intsalled, using the register, benchmark and downloared
I'd rather put
Wait(3600) than Hangup(). Furthermore hangup would probably not work
since the line was not taken offhook.
Asterisk would do cleanup on the logical zap channel but then the next
ring would create another zap channel and so on till the line stops
ringing.
Martin
On Thu, Apr 2
Then you need to edit /etc/dahdi/system.conf
manually and add
fxsks=1
then dahdi_cfg -vv
then check if wcfxo module takes interrupts
dahdi_test
Martin
On Thu, Apr 2, 2009 at 5:55 PM, Manolet Gmail mano...@gmail.com wrote:
What's the output of:
lsmod | grep ^dahdi
r...@lhserver
it out ... this is the first step...
dahdi_cfg -vv should show all your 64 channels
Martin
On Thu, Apr 2, 2009 at 6:36 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello!
I am trying to configure my digium TE220 dual-span pci express card
with Dahdi. I seemed to have managed to set up the card
is your agent configured to support call waiting ? if so then THIS can happen
Martin
On Thu, Apr 2, 2009 at 3:57 PM, David @ULC ucoms2...@gmail.com wrote:
Many time we face an issue where even if an agent is on Call, another call
comes in.
Sometimes, even if agent hang up the call, call
the channels
Martin
On Thu, Apr 2, 2009 at 7:18 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello!
Here is all I got:
system.info:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
dahdi_channels.conf:
;This context
make it asterisk -vvvc (CONSOLE MODE)
On Thu, Apr 2, 2009 at 7:36 PM, Martin asteriskl...@callthem.info wrote:
ok,
1) you're missing switchtype=euroisdn ...
2)
so edit /etc/asteirsk/logger.conf make sure console = is not
commented out; if it is then uncomment
service asterisk stop
... and then
get the trademark and then prove it in court ...
seriously - no normal country will trademark two words out of the
regular vocabulary
Martin
On Thu, Apr 2, 2009 at 6:39 AM, Gabriel - IP Guys
gabr...@impactteachers.com wrote:
Dear All,
Thanks for taking the time to read this. I have
It's all in the
CLIshow application dial
and read what it has to say ...
-= Info about application 'Dial' =-
[Synopsis]
Place a call and connect to the current channel
[Description]
Dial(Technology/resource[Tech2/resource2...]
^^
Martin
You mean when the driver is not loaded ?
It doesn't. The driver enables the current drawn.
Well that is my guess. But since I have one card handy I'll confirm for you.
CONFIRMED. No power without the driver loaded
Martin
On Thu, Apr 2, 2009 at 4:38 PM, Noah Miller noahisaacmil...@gmail.com
... UNLESS
for some reason you only compiled
chan_dahdi.so and copied it manually leaving the old app_meetme.so
with HAVE_ZAPTEL flag...
paste your lsmod output
Martin
On Thu, Apr 2, 2009 at 12:22 PM, Dave Poirier dpoir...@mesd.k12.or.us wrote:
We recently updated our Asterisk (1.4.24) box from
to trademark something composed of two words and they
refused. So it seems I know what I'm talking about.
Martin
On Thu, Apr 2, 2009 at 7:54 PM, Cary Fitch ca...@usawide.net wrote:
Heck,
There goes
General Electric
General Motors
Headline News
General Dynamics
General Instruments
All
repeat it ... You might also check whether there's
a firewall somewhere and the RTP session can work properly ...
You'd have to inspect the UDP ports mentioned in the SDP of INVITE and
200 OK to INVITE (turn on sip debug on asteirsk CLI to catch the SIP
messages)
Martin
On Thu, Apr 2, 2009 at 2
in zaptel/dahdi it should be zapata.conf/dahdi.conf transfer=yes
and/or callwaiting=yes
as far as I know with SIP devices it's usually configured on the
device itself, though you could limit it I guess
in the definition of the sip account ... don't remember the keywords,
google it out
Martin
is_context_set that would be NULL if no context keyword is processed
from the sip.conf etc.
That is easier to check instead of comparing against default_context
Martin
On Wed, Apr 1, 2009 at 2:45 PM, Anthony Plack t...@plack.net wrote:
Okay, I am not understanding if I have this correct or not.
I have
I wonder why people don't get it ? X100P is a winmodem was and always will be.
Martin
On Wed, Apr 1, 2009 at 12:26 PM, Tim Nelson tnel...@rockbochs.com wrote:
If the primary purpose is to drive down cost, why not simply buy any one of
the existing 'Wildcard X100P' clone cards
Dear All,
Is anyone having luck with using some code for SIP network topology
hiding + NAT traversal (SBC functionality) with Asterisk ?
I tried OpenSBC but it didn't do NAT from Asterisk to ATA correctly.
It's in plans for OpenSIPS but it's not implemented yet ... checked
their svn.
Martin
knows
the exact numbers since it's dependant on your kernel version/asterisk
version/CPU/motherboard/ethernet card/
memory speed/hdd speed etc.
Just make sure the message is encoded in G711 ulaw/alaw so there's
no transcoding... (use sox)
Martin
On Wed, Apr 1, 2009 at 10:46 PM, Erick Perez eaper
Am Freitag, den 27.03.2009, 16:35 + schrieb Phil Reynolds:
Quoting Chris Bagnall li...@minotaur.cc:
Thins number is wrong - it has too many digits - should only be eight
after the 20. (possible you put a surplus 3 in?)
Good guess, indeed +44 20 3393 7389 has an answering machine as
Martin
On Wed, Mar 25, 2009 at 9:40 AM, David Anthony O Reilly oreil...@tcd.iewrote:
Hi all
I have been hacked but no idea how!!! I noticed somebody in Eastern Europe
came from an American IP and tried to call loads of international numbers.
Thankfully I had no credit with my VOIP out provider
I wonder why they only allow G.729 with this ... where's the great sound of
the skype call now ?
Martin
On Mon, Mar 23, 2009 at 2:42 PM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
Anyone connected up to it yet?
http://www.skypeforsip.com/
It would
Are you bulding it from rpmbuild ?
The error says it can't find the asterisk.h so it's most likely a
Makefile/paths error.
go to readline.c to where it's trying to #include asterisk.h and fix it
there :)
Martin
On Sat, Mar 21, 2009 at 11:06 AM, Frederik Himpe fhi...@telenet.be wrote:
On Sat
for callback agents)
Are you using callback agents ? Can you describe the Queues announcement
problem in more detail ?
Martin
On Fri, Mar 20, 2009 at 1:05 PM, Cary Fitch ca...@usawide.net wrote:
[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device
state of this queue member, SIP
the incoming audio
frame. So check if the Answer + Echo application work.
Also if you have VAD which means your SIP device doesn't send audio frames
when there's silence detected =
you'd also have the same problem = not see audio going to your phone.
Martin
On Fri, Mar 20, 2009 at 6:07 PM, Greg Hinson
= _1NXXNXX,1,Goto(+${EXTEN},1)
exten = _011.,1,Goto(+${EXTEN:3},1)
;USA
exten = _+1NXXNXX,1,Answer()
exten = _+1NXXNXX,n,Macro(enumdial,${EXTEN})
exten = _+1NXXNXX,n,Set(CALLERID(num)=+18579284409)
exten = _+1NXXNXX,n,Playback(pls-hold-while-try)
and so on...
Martin
An example of how
##switchtype=national
##pridialplan=national
channel=1
Where is channel 2 configured?
channel=1-2
signalling=fxs_ks
language=us
context=line-1
group=0
##switchtype=national
##pridialplan=national
channel=4
channel=3-4
Martin
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to another (but more than 20yrs ago...)
Martin
11xx is used for the rotary dial equivilant of *xx on many central
office switches.
Assuming you are not using rotary dial, I generally use 4 digit
extensions with the 11xx format for the same reason you suggest.
--Shane
outlets coming from two opposite ends of the
building can cause real headache!) Check it first. Then you may want to
continue what your former IT admin tried to start :-)
Martin
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Carlos Chavez wrote:
Use the h extension and execute DeadAGI.
Seems to be working. I have access to variables too.
David fire wrote:
you can try whit the g option to dial.
David
This works only when the called side hungs up, but not the when caller
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer, busy, etc., it is not a
problem, because dialplan continues after the Dial() application, but
when the call is
Hi Robert all,
Maybe someone else can speak to using Progress(), but I don't know if it
is required or not. On our system, we didn't need it, and these settings
below (plus a call to the telco to tell them to turn on operator
messages, don't eat them) did the trick.
Good luck,
Martin Smith
are sometimes easier to reference against the cause codes.
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
to HANGUPCAUSE.
Good luck,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Boardman
Sent: Friday, November
could
still rename the file AFTER the Dial is complete just using System().
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL
to
another location.
I'm not sure how else to describe what the user wants to do, but I'm
willing to try if people have questions :)
Is there a simple way to do this without a meetme room?
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University
to the
addCommand method, and there's some explanation of the fields.
Hope that helps,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
to your daily digest e-mail if you want a new
post. Some threading newsreaders might not like that, or so I've heard.
But you're doing great so far; good luck!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
Am Dienstag, den 07.10.2008, 01:42 -0700 schrieb Vieri:
Hi,
Currently I'm using Asterisk 1.2 and 1.4 in different setups. When a user
wants to pick up a call
within his/her pickup group, *8 must be dialed (or whatever you define in
features.conf).
[...]
I was thinking of configuring some
.
That was indeed the problem. I added this to iax.conf:
[myprovider]
type=friend
username=88821268
secret=xxzzyy
host=s1.core.myprovid.er
And used this in extensions.conf:
exten = _ZXXX,2,Dial(IAX2/myprovider/${EXTEN:0},30,r)
Thank you for the assistance.
Regards,
Martin Seebach
a timeout with no digits pressed. I'd also
encourage you to check out the Asterisk-Java mailing list via
http://asterisk-java.org/development/mail-lists.html.
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171
working fine for a while, and others report IAX2
working fine.
- Martin
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users
) 0
1 modules loaded
Did you build and load ztdummy (assuming you have no Zaptel/Dahdi cards?
No - but i don't use MeetMe?
Thanks,
Martin
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debugging info? I can't figure out what's
wrong.
Thanks!
- Martin
( my iax.conf and extensions.conf on http://pastebin.com/mb0020bd )
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Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 705-1400
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,log,verbose,command,agent,user
Le lundi 22 septembre 2008 ? 09:46 -0400, Jason Martin a ?crit :
Hello,
I'm trying to do some simple tests with AJAM on asterisk 1.4.17 and I'm not
having much success.
Right now the http server just listens on localhost:8088. I've used lynx and
elinks
/tiki-index.php?page=check_asterisk.
Cheers all,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dean Collins
Am Freitag, den 29.08.2008, 09:16 -0700 schrieb Ira:
At 05:48 AM 8/29/2008, you wrote:
(so since they still liked the Snoms otherwise, my solution is to get them
to dial a star at the end of a number to select their 'home' account,
otherwise it goes out on their work account and the dialplan
Hi Stefan,
I'd expect there's a Manager event that is fired when an IAX client
login happens. You could watch for that and initiate your call if
there's voicemail at that time.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352
That's a good point. I don't know, honestly, if you can react to a SIP
register or an IAX login from within the dialplan. To anyone else:
Is there a way to act in the dialplan on a manager event?
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
/mail-lists.html.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 20
Am Dienstag, den 19.08.2008, 02:53 + schrieb Miguel Otamendi:
Please, I need help.
I have problem witch voicemail.
-- Executing [EMAIL PROTECTED]:3] VoiceMail(Zap/4-1, s) in new stack
[Aug 17 21:33:46] WARNING[11864]: app_voicemail.c:3061
leave_voicemail: No entry in voicemail
Here's my attempt to explain a quick way of doing an auto dialer with
Scala and the Asterisk-Java library:
http://blogs.reucon.com/asterisk-java/2008/08/10/outbound_message_delive
ry_using_agi_and_ami_in_scala.html
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic
Am Dienstag, den 22.07.2008, 14:53 -0500 schrieb Gregory Malsack:
Hello,
My boss is asking me to setup the asterisk server to record all calls.
(Simple). However, he wants to be able to enter a key sequence during
the call to stop the recording. Any ideas on how I would do that?
Hi
with asterisk, as asterisk needs to know if it
should provide any services for the call (music on hold, transfer,
etc).
yes, 'only' rtp goes direct, SIP stay on asterisk since it might
be a hangup or something else comes in.
Yours
Martin
Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia:
Need to have a different TONE for any internal call (EXT OR TRANSFER)
from an external (outside) call.
Any suggestions?
Fidel,
I do not know what kind of tone you mean:
The sound of a phone that signals a call coming from
Am Montag, den 14.07.2008, 09:45 -0700 schrieb bilal ghayyad:
Hi All;
Anyone can advise for a method to have open vpn client to be installed on the
mobile, so it can open a vpn channel with Asterisk (I installed open vpn at
it) from the mobile, and then I can let fring use the open vpn
Martin
- Original Message -
From: Martin Schrott - thinking:systems
To: [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial
Discussion
Sent: Tuesday, July 01, 2008 6:18 AM
Subject: Re: [asterisk-users] queue welcome message
Hello Tarek,
thank you for your
for the problem to appear.
What's going on? How can I fix this? Where should I look?
Thanks,
--
martin; (greetings from the heart of the sun.)
\ echo mailto: !#^.*|tr * mailto:; [EMAIL PROTECTED]
with sufficient thrust, pigs fly just fine. however, this is not
necessarily a good idea
is not the first
in line.
specifiing a periodic announce does play the message after the
periodic-announce-frequency has been over.
Is there also something else we can use?
Thank you
Martin
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Hello Tarek,
thank you for your idea. But this only would work for the first caller - when
the moh starts.
all other callers go directly into moh on the position where the first caller
is in moh.
So this does not work. :-(
Anyone an other idea?
thank you
Martin
- Original
Am Samstag, den 28.06.2008, 08:15 -0500 schrieb [EMAIL PROTECTED]:
Hi List
I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has already
processed more than 10million calls!
I have one big challenge which is reporting... it is the requirement to
have a web reporting module which
you see, everything is possible.
hope to help,
Martin
- Original Message -
From: Thomas Winter [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 24, 2008 11:22 AM
Subject: [asterisk-users] Queue with different music for each caller
Hi,
is there an possibilty
Hello Thomas,
no problem.
In asterisk 1.6 use
SetMusicOnHold(musiconholdname)
then it will work in older Asterisk versions!
br,
Martin
- Original Message -
From: Thomas Winter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
setting MONITOR_OPTIONS=b
Can anybody tell us, how we can get that running?
We would like to call something like:
/usr/bin/xyz -a -b -c
Thank you very much,
Martin
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on and hangs up. :-(
we triyed to use the c flag
and the timeoutrestart
both did not work.
How could we set up the queue to go on after a call?
Hope anybody can help.
thank you
Martin
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] DTMF[11028] channel.c: DTMF end '1' received on
SIP/199-b31ddc00, duration 80 ms
[Jun 9 16:26:21] DTMF[11028] channel.c: DTMF end '1' put into dtmf
queue on SIP/199-b31ddc00
Thanks!
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
[12300] chan_zap.c: Ending VLDTMF digit '2'
[Jun 9 16:47:55] DEBUG[12300] chan_zap.c: Started VLDTMF digit '1'
[Jun 9 16:47:56] DEBUG[12300] chan_zap.c: Ending VLDTMF digit '1'
Thanks :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University
diaplan designer or visualizer. The web start demo requires Java 6.
I'd love your feedback.
Thanks,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
) but I get an Auto fall through
message and it just rings out.
What have I got wrong? I just want an easy way to match two sets of
numbers 'owners' and 'friends' all other callers should hit the last
'call-house' jump.
TIA
Martin
[globals]
house-numbers=SIP/officeSIP/lounge
my-mobile-number
Have you tried GetVariableCommand and GetFullVariableCommand?
See
http://asterisk-java.org/development/apidocs/org/asteriskjava/fastagi/co
mmand/GetFullVariableCommand.html.
Martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sherwood McGowan
,
=Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Gibson
Sent: Friday, April 18, 2008
time -- in which case a Java daemon may even be able to outperform more
traditional languages as it optimizes at runtime :).
Cheers,
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
, critical, alive, host,
printer, allow(ed), cpu
I'm sure whatever we end up with will be useful though, so thanks either
way! :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message
Perhaps a Talk like a pirate day prompt! :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Johansson Olle
.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Nunez
Sent: Friday, May 02, 2008 3:51 PM
To: 'Asterisk Users
:
http://www.aharef.info/2006/05/websites_as_graphs.htm
More soon :)
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
From: [EMAIL PROTECTED]
[mailto:[EMAIL
-commerical list like this one.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Salvatore Giudice
Sent: Friday
and clean out of sand and drywall pieces :-(
Martin
- Original Message -
From: Justin Newman
To: asterisk-users@lists.digium.com
Cc: [EMAIL PROTECTED]
Sent: 11. dubna 2008 13:00
Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
Did this just start
Driver no yes no
MGCP Media Gateway Control Protocol (MGCP) yes yes no
Agent Call Agent Proxy Channel yes yes no
--
7 channel drivers registered.
*CLI module reload chan_zap.so
No such module 'chan_zap.so'
*CLI
--
Murithi Martin
NOTICE: The contents of this e-mail and any accompanying
/CiscoPhone
Context: sip
Priority: 1
Timeout: 3
CallerID: Martin Edlman 38
Exten: +420phonenumber
- --
Ragards,
Martin Edlman
Fortech, spol. s r.o,
Ropkova 51, 57001 Litomyšl
Public GPG key: http://edas.visaci.cz/#gpgkeys
-BEGIN PGP SIGNATURE
rejection for user martin f. krafft
sip:[EMAIL PROTECTED];tag=fipzt
and SIP debugging then prints:
OPTIONS sip:sip05.insphone.ch SIP/2.0
Via: SIP/2.0/UDP 84.75.148.xxx:5060;branch=z9hG4bK71785803;rport
From: asterisk sip:[EMAIL PROTECTED];tag=as05fc20f4
I am not calling as username asterisk
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