yes start it with asterisk -gc
watch and see what the error is.
bkw
On Wed, 20 Aug 2003, Marcus Adolfsson wrote:
I upgraded to the latest CVS yesterday (and this morning again), and
whenever I execute the reload command Asterisk seems to hang. While the
current calls aren't
You can also check www.openh323.org/bin/
bkw
On Wed, 20 Aug 2003, Steven Thomas wrote:
Thanks - because of my ignorance using the CVS archive - could you please
give me the full command - thanks.
Regards,
Steven Thomas
Chee Foong
astman or gastman would tell you this info. And yes we us it in
production right now. Works better than anything we have had previously.
bkw
On Wed, 20 Aug 2003, Ernest W. Lessenger wrote:
At 10:42 AM 8/20/2003 -0500, you wrote:
I've literally read the last year's worth of posts to
VAD is evil. I hate it. I find when we used it.. you keep asking people
to repeat stuff all the time.. and it was just anoying.
bkw
On Wed, 20 Aug 2003, WipeOut . wrote:
Does the Asterisk server support VAD (aka Silence Suppression)? I want my SIP
phones (7960's) to use VAD when dialing
I would use the latest CVS for one. And try again.
bkw
On Wed, 20 Aug 2003, Ian Blenke wrote:
I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one
another using IAX/IAX2 trunks.
I've managed to get a semi-functional NAT Firewall working as a PBX
(with Asterisk running
Or you can jump on #asterisk
bkw
On Wed, 20 Aug 2003, John Brown wrote:
We are getting ready to replace our old Panasonic PBX with
an Asterisk system. I'd say its ready for prime time.
THe other thing is to have a good consultant in your back pocket
for those now how do I do this.
I can
RCF3389 defines Payload for Comfort Noise, that is used with VAD.
So I turned it off on my endpoints (ATA186 and c827-4v)
Eduardo
On Wed, 20 Aug 2003 11:35:14 -0500 (CDT)
Brian West [EMAIL PROTECTED] wrote:
VAD is evil. I hate it. I find when we used it.. you keep asking
Does http://www.voicepulse.com/ work with *?
On Wed, 20 Aug 2003, John Todd wrote:
At 3:20 PM -0500 8/20/03, Mike Ciholas wrote:
Hi all,
While pondering my choices for local dial tone service via a
bunch of POTS lines or a T1, I began to wonder if perhaps there
is another way.
Are
pipe my local CO line into my * box with an X100P
bkw
On Wed, 20 Aug 2003, Mike Ciholas wrote:
On Wed, 20 Aug 2003, Brian West wrote:
I think NuFone can do what you need contact [EMAIL PROTECTED]
I have inbound 800 service and outbound ld service with them..
works great
Seeing your PBX go down in flames due to bad hardware support...
PRICELESS!
On Thu, 21 Aug 2003, Anthony Wood wrote:
Here are some options:
Digium X100P x 4
US$100 * 4 = US$400
well supported by asterisk
manufacturer supports asterisk developers
Deployed in lots of
Or as I like to call it... LookOut Express... because if you run it.. you
better LookOut.
bkw
On Wed, 20 Aug 2003, Jon Pounder wrote:
Here's a better tip - don't use MS outhouse that likes to open attachments
for you.
Oh yeah, that's a feature not a bug, I forgot.
At 08:29 PM 8/20/2003
Jeremy,
That is one of the reasons I have went with nufone and the company
I work for is going to be going with nufone. No need having idle channels
when someone could be putting them to use! :P (and you make $$) Everyone's
happy.
bkw
On Wed, 20 Aug 2003, Jeremy McNamara wrote:
Dan
If you are using chan_h323 you need to read the src.
noFastStart
noH245Tunneling
noSilenceSuppression
Those are just some of the options I see in the src.
bkw
On Tue, 19 Aug 2003, Langley, Sean wrote:
In my h323.conf file I have put in the line
faststart=disable
But when I sniff packets
Well here is a little bit of news.. I bet we have all heard this before.
Wishful thinking? ;)
bkw
-- Forwarded message --
Date: Tue, 19 Aug 2003 13:43:33 -0400
From: David Li [EMAIL PROTECTED]
To: Brian West [EMAIL PROTECTED]
Subject: RE: IAX, Asterisk, GSM,SPEEX and ILBC
put mpg123 in /usr/bin
bkw
On Tue, 19 Aug 2003, Asterisk - linux - JVB wrote:
Yes I linked all the mp3 and mpg extensions with the mpg123 program
(/usr/local/bin) ... but still not able to get the music on hold playing
Getting curious now what I am doing wrong ...
Andrew Joakimsen wrote:
Good luck.. I toasted one over the weekend that was locked also.
On Tue, 19 Aug 2003, John Todd wrote:
If anyone out there has an ATA-186 that they purchased but cannot use
with Asterisk due to it's being locked by Vonage, please contact me
off-list.
JT
On Wed, 20 Aug 2003, Jamie Carl wrote:
Seeing as no one else has replied, I figured I may give it
a shot. At least it'll start something.
Now, correct me if I'm wrong someone, but as far as I
understand in this situation you can do both. Normally
the RTP packets would be swtiched
Let me try this once again. :P The reason I wanted everything to go thru
the * server is so you can monitor calls with res_monitor.
bkw
On Wed, 20 Aug 2003, Jamie Carl wrote:
Seeing as no one else has replied, I figured I may give it
a shot. At least it'll start something.
Now, correct
You are almost there... http://www.loligo.com/asterisk/current/
Check that.. see how he has it setup... you have a few things in this
config that will cause it to not work correctly.
bkw
On Wed, 20 Aug 2003, Yehiel Samson wrote:
I have a small HUGE problem with *.
I have installed * but I
I see all these posts about wanting a script for prepaid setup... Have you
not even tried to look it up or put any effort forth? If you stop and
think about it its not that hard. It takes alot of error checking, alot
of testing to make sure it does correctly. I did something simple today
just
http://www.bkw.org/~brian/agi-ccard.agi
Something a bit more complex.. using cdr_mysql and DBI... It needs to be
re-written totally from ground up .. proof of concept.
bkw
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[EMAIL PROTECTED]
I use it without issues.
[agentlogin]
exten = 800,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM})
exten = 800,2,Playback(agent-loginok)
exten = 800,3,Hangup
exten = 801,1,RemoveQueueMember(techsupport|SIP/${CALLERIDNUM})
exten = 801,2,Playback(agent-loggedoff)
exten = 801,3,Hangup
Our device
of a call are
unsynchronized by the duration of the interval that it takes for the answering leg
to pick up the phone. This can be very distracting in a final mixed version of
the file.
Brian West ([EMAIL PROTECTED]) came up with a clever solution to this. Since we
know the ENDING
Maybe we can pester kram to make that an option. monitor.conf anyone?
bkw
On Mon, 18 Aug 2003, John Todd wrote:
So how does one emit the legally required ( in some locales)
10 to 30 sec soft beep, letting people know they are being recorded ??
very cool trick using the end point as the
Go for it! :)
On Mon, 18 Aug 2003, Mark Spencer wrote:
Does anybody have any reason why I should *not* permenantly replace
app_voicemail with app_voicemail2? If so, speak now or forever cvs update
-D 8/18/2003.
Mark
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Asterisk-Users mailing
at 14:59, Brian West wrote:
Maybe we can pester kram to make that an option. monitor.conf anyone?
bkw
Well, while we're in the let's pester Mark mood... why not have him
fix res_monitor so it writes to just one file! That would sure make me
a lot happier...
Jared Smith
Just alias the commands.
On Mon, 18 Aug 2003, Tilghman Lesher wrote:
On Monday 18 August 2003 04:06 pm, Mark Spencer wrote:
Does anybody have any reason why I should *not* permenantly replace
app_voicemail with app_voicemail2? If so, speak now or forever cvs
update -D 8/18/2003.
I agree with jtodd on that one it would make life simpler.. I don't
care if the files are seperate or not.. thats an easy solution to
overcome.
bkw
On Mon, 18 Aug 2003, John Todd wrote:
On Mon, 2003-08-18 at 14:59, Brian West wrote:
Maybe we can pester kram to make that an option
Works for me.. I can press # and dial the ext and press # to transfer a
call.
www.bkw.org/~brian/ata.html for the settings I used in my ATA
bkw
On Mon, 18 Aug 2003, ASN wrote:
Hi all:
I'm testing a new installation of *, bringing up some ATA186. In * environment, all
stuff works greats.
:12 -0600, Jared Smith wrote
On Mon, 2003-08-18 at 14:59, Brian West wrote:
Maybe we can pester kram to make that an option. monitor.conf anyone?
bkw
Well, while we're in the let's pester Mark mood... why not
have him fix res_monitor so it writes to just one file!
That would sure
Have you tried to CVS recently? I had some similar issues and mark fixed
those.. but I wonder if they could be related to this.
bkw
On Sat, 16 Aug 2003, John Brown wrote:
Hi list,
I've got a testbed running with the following config:
1. RH 7.3 linux machine
2. 2 Grandstream phones
3. 2
Repeat after me. Telnet BAD ssh good!
bkw
On Fri, 15 Aug 2003, Andy Powell wrote:
Personally I'd use ssh rather than telnet
Andy
*** REPLY SEPARATOR ***
On 15/08/2003 at 12:21 Steve Lane wrote:
I am having problems trying to run asterisk from a telnet session. I
http://www.digium.com/index.php?menu=asterisk_g729
bwk
On Fri, 15 Aug 2003 [EMAIL PROTECTED] wrote:
Hi
I am using asterisk with a Quicknet lineJack card. I am trying to get a
proof of concept demo together before real deployment. I have a couple of
qustions.
1) Can I use the codecs that
Oh my why do that? Customers/Users will have a hard time hearing and
understanding in some cases.
bkw
On Wed, 13 Aug 2003, Stuart Hirst wrote:
Does anyone know if it would be possible to play music on hold in the
background whilst playing IVR prompts. I am hoping that this would have
the
http://www.bkw.org/~brian/ata.html
Pay attention to connectmode and audiomode Its how I set it up and it
works.
bkw
On Thu, 14 Aug 2003, Dan wrote:
Hi Brian,
ATA is in SIP mode, and RFC2833 is used.
Something else to check?
Thanks,
Dan
- Original Message -
From: Brian
to
agentA regarding 'INHOUSE' policies, and how it will effect the agents
employment!
G
Brian West wrote:
But how do you translate inhouse to logic for app_queue. :P
On Mon, 11 Aug 2003, Richard Lyman wrote:
ok and what happens when agentA in on a 3 hour call? once again i think
this type
effectively.
i know there is at least one other out there that agrees with
me. G
Brian West wrote:
I was speaking in a logic related to real call routing and queueing. In
House policy can be built on top of your call strategy. What we are
needing is input on logic only ..
bkw
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934031360.htm
Thats a good start..
bkw
On Tue, 12 Aug 2003, Fabrice Tereszkiewicz wrote:
Hello,
I would like to buy a SIP IP phone, but I don't know wich one to
choose... Can you tell me wich IP Phone is known to work with
Ya what he said! :P
On Thu, 14 Aug 2003, Martin Pycko wrote:
And remember to use Dial with t option
Martin
On Thu, 14 Aug 2003, Brian West wrote:
http://www.bkw.org/~brian/ata.html
Pay attention to connectmode and audiomode Its how I set it up and it
works.
bkw
On Thu
' the agent you want first
then if fail they go right to the next one in the 'ordered' list.
Brian West wrote:
leastrecent suffers the same fait as fewestcalls onlying ringing the
leastrecent agent over and over endlessly. It should have a fallback
option.
roundrobin with leastrecent
I used the exact versions listed in the readme for chan_h323 and it works
fine. Slackware and RH8 and 9.
bkw
On Fri, 8 Aug 2003, Kelvin Chua wrote:
hi guys,
i'm encountering one way audio on cvs using netmeeting and chan_h323.so
is there a quick fix or workaround for this?
compiled using
While we are on this subject. For testing and such I have been trying to
get one asterisk server to register with another via sip(i know i know use
IAX) but it doesn't work It should... I can't see any reason it
shouldn't Any pointers? All I get is proxy auth and * crashes a
bloody
as I can tell,
it doesn't. The standard queue/agent logic requires that you assign an
extension to a phone.
Someone correct me if I'm wrong, please. :)
- Devon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian West
Sent: Wednesday, August 13, 2003
AgentLoginCallback does this doesn't it?
On Wed, 13 Aug 2003, Steven Critchfield wrote:
On Wed, 2003-08-13 at 11:28, Devon Henderson wrote:
We're still in the planning stages of our Asterisk implementation, but we
have a requirement that the extension be mapped to a user, with the phone as
Correct me if i'm wrong but doesn't the cdr modules log the call duration?
bkw
On Wed, 13 Aug 2003, Alastair Maw wrote:
Hi.
I'm trying to do some custom call logging, and I want to call an AGI
script from a hangup handler to log call durations and things. Although
the script executes, it
it rotate?
Will the same person always be the first and the order thereafter always the
same?
W
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, August 09, 2003 8:06 PM
Subject: [Asterisk-Users] app_queue, fewestcalls and leastrecent logic
Call waiting should do that.. i'm not sure exactly how it would work in
your situation.. I know we disabled that anoying beep on our ATA's because
it was very distracting. We queue callers with ringall strategy right
now. Going to switch to roundrobin next week or so.
bkw
On Mon, 11 Aug 2003,
Welcome to the club... I can't get it working either.
bkw
On Wed, 13 Aug 2003, Eric Wieling wrote:
Is there any way on the iaxtel.com web site to see if my asterisk is
registering and what 700 number is associated with my iaxtel account? I
registered many months ago but never used it. My
did you happen to run ztcfg after you setup the configs?
On Sat, 9 Aug 2003, Barry Porch wrote:
I am attempting to set up an Asterisk box which I am only concerned with
getting a single T1 working. I have this T1 connected to my PBX and I
am looking at using Asterisk as a conference bridge.
First of all I would like to thank Mark for getting roundrobin to go
roundrobin. Good job.
Now we have some options here for leastrecent and fewestcalls strategy. It
needs some work on the logic and Mark recommend that I ask the list and
get some input before he makes any changes to it.
Does asterisk work with Vonage? I see all this talk or are you guys just
plugging it into an FXO port?
bkw
On Wed, 6 Aug 2003, Steve Meyers wrote:
On Wed, 2003-08-06 at 13:39, John Schmerold wrote:
I've canceled my Vonage service because of the requirement to prefix
every call with a 1.
first
then if fail they go right to the next one in the 'ordered' list.
Brian West wrote:
leastrecent suffers the same fait as fewestcalls onlying ringing the
leastrecent agent over and over endlessly. It should have a fallback
option.
roundrobin with leastrecent first
roundrobin
Apparently you didn't read the README.. Please read that over again.. it
tells you exactly what to do.
bkw
On Sun, 10 Aug 2003, Serge Mankovski wrote:
Hi
I am using inAccess channel driver.
Compiled, installed. This is what I get when I am trying to start *
Anyone have any luck setting up festival 1.4.3? Can someone share some
input.
bkw
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If I understand this correctly.. Your channel bank will convert analog FXO
FXS to a T1 then you slap a Wildcard T100P in your linux box then your
all set. It shouldn't matter if the channel bank is compatible with linux
or not because you are going to terminate with a T1 cross over to your
linux
Does your IOS on the AS5300 support g711ulaw?
bkw
On Mon, 4 Aug 2003, Luciano Ramos wrote:
I've tested g711ulaw and alaw in cisco AS5300 and they don't work
with asterisk, gsmfr in AS5350 works great.
Luciano
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Sat, 2 Aug 2003, Brian West wrote:
I have figured out that its a problem in app_queue, could be the
interaction between chan_sip and app_queue. Or the ATA is on crack.
in chan_sip if I change
case 501: /* Not Implemented */
if (owner)
ast_queue_control(p-owner
Everyone does now.. I don't get it.. they have a product we want.. but
they wont or can't sell it. Guess they can't keep up with demand right
now.
bkw
On Sat, 2 Aug 2003, Steven Honson wrote:
I get a username/password prompt when I go to that page...
This is what Brian West at Thu, Jul 31
I'm currently working on a perl script convert csv logs to a http log
equiv:
LogFormat %h %l %u %t \%r\ %s %b \%{Referer}i\ \%{User-agent}i\
Right now I have output something similar to this:
111 - - [02/Aug/2003:16:39:15 -0500] GET /300 HTTP/1.0 200 6144 sipext ANSWERED
INCOMING - -
I noticed a few issues with app_queue just wanted to know if its sip
related or ata186 related:
Ext 111 and Ext 112 are dynamically loged into the queue via
AddQueueMember.
Call hits queue with fewestcalls routing.
Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some
reason ext
the
next time that extension comes around. So is it a bug with the ATA? Or
app_queue or the interaction of app_queue and chan_sip
ATA Version: v2.15 ata18x (Build 020927a)
bkw
On Sat, 2 Aug 2003, Brian West wrote:
I noticed a few issues with app_queue just wanted to know if its sip
related
Check application DISA: DISA (Direct Inward System Access)
bkw
On Fri, 1 Aug 2003, Michael Baird wrote:
I've designed a voice menu, someone calls a certain extension, and I
send them to another context via a goto, and play a background message.
After playing this message can I provide them
Figured out what to do.
exten = 900,1,Queue(techsupport|HTt)
exten = 900,2,Voicemail2(b111)
And DO NOT put members = in the queues.conf Let members come and go
via the AddQueueMember and RemoveQueueMember
Works perfect.
bkw
On Thu, 31 Jul 2003, Brian West wrote:
I currently have
I was told in #asterisk that you just hit transfer, dial the extension,
speak to caller and press transfer once your done talking and it should do
it. In addition you can do transfer+extension+transfer+hangup...
Thats how I was told it would work.
bkw
On Wed, 30 Jul 2003, denon wrote:
Last
Same here. Same build.
On Wed, 30 Jul 2003, Dan wrote:
Hi Erik,
I have the version X-Lite 2.0 private build 1050.
When I click on transfer then extension then transfer, the call is closed,
but the final destination does not ring.
Thanks,
Dan
- Original Message -
From: Erik
I have this setup:
Sip Phones - Asterisk - h323 gateway - ptsn
Sip phones are setup for out of band dtmf
but the h323 gateway is inband. Is their a way to pass the digits from
the sip phones to the ptsn via the h323 gateway?
bkw
___
Asterisk-Users
=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
I have this setup:
Sip Phones - Asterisk - h323 gateway - ptsn
Sip phones are setup for out of band dtmf
but the h323 gateway is inband. Is their a way to pass
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
I have this setup:
Sip Phones - Asterisk - h323 gateway - ptsn
Sip phones are setup for out of band dtmf
but the h323 gateway is inband. Is their a way to pass
DTMF.
On Wed, 2003-07-30 at 15:26, Patrick wrote:
I have the same setup, and in the sip.conf file I set the dtmfmode=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
I have this setup:
Sip Phones
I would love to see FreeBSD support. Any links on the OpenBSD port?
bkw
On Tue, 29 Jul 2003, Tilghman Lesher wrote:
On Tuesday 29 July 2003 09:41, Troy Settle wrote:
-Original Message-
From: Low, Adam
Sent: Tuesday, July 29, 2003 9:15 AM
Personally, I've compiled
, 29 Jul 2003, Brian West wrote:
I have put together this call forwarding and dnd config:
I'm sure it can be dome with macro's but I couldn't figure that out...
anyone care to input.
74 Turns DND on my phone will not ring, drops caller to voicemail...
73 Turns DND off
72+ext forward your
Here are a few things I would like to see ..
1. In addition to time/date stamps, store/read the caller id info with the
voicemail messages.
2. Have the ability to configure the system to ignore and delete messages
left by a caller that are 3 seconds or less (maybe make this configurable)
Not
Does anyone have any hunt group examples?
phone 1
phone 2
phone 3
message press 1 to leave a message
loop back to phone 1
till the call is answered...
Any examples?
Thanks,
Brian
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Ok I have festival on RH8. It speaks fast and you can't understand it. I
don't have any FXO cards in this box yet. Can someone shed some light on
this?
Thanks,
Brian
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I would also like to see a patch to ignore voicemail messages x number of
seconds long.. ususally those 1-4 second voicemails are nothing anyway..
bkw
On Sun, 27 Jul 2003, Tilghman Lesher wrote:
On Saturday 26 July 2003 21:06, Andy Hester wrote:
Tilghman,
I applied your
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