quote who=Matt Fredrickson
On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote:
Yes, but what would one do there?
One who doesn't gamble, drink, or carouse, that is.
I am making my first trip to LV later this Fall, and I dread it.
I can't imagine what I'll be able to find to do
could not make headway on the first look.
--Rob
--
Robert G. Ristroph
Airlink Systems
[EMAIL PROTECTED]
(512) 231-1240 x103
This message was sent using IMP, the Internet Messaging Program
I am going to be traveling and I wanted to be able to get on the
internet and call thru * to make calls. The problem is I do not have a
fixed ip address. How do you make this work? I will be using IAXCOMM. TIA
Bob
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t=incoming---end
zapata.conf
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Robert
WagnerSent: Wednesday, September 14, 2005 10:46 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] TE110P -
Title: TE110P - [EMAIL PROTECTED] Install Problems
I am having problems sending and receiving calls over the T1. They never seem to connect - outbound keeps ringing, inbound gets busy. The T1 looks ok - no errors on the line. Any ideas on what is wrong? I have tried a variety of fxsks and
Maybe this could be used with the Internet Repeater trunking system I
primarily use VHF... But would be interested in setting that up on my asterisk
with the Internet 2M Repeater trunking system inter-connect
Robert A. Huddleston, KF4BYY
Cavalier Telephone LLC.
(Desk) 804.422.4401
(Cell
itself.
Its rather hackish (it uses a
global context not linked in with the regular context list), and so probably
has some issues, but I can clean it up and post the patch somewhere if others
are interested. It sounds like you would be.
Cheers!
Robert Bedell
From:
[EMAIL
company is boosted by X percentage??
TIA,
Robert
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appreciated.
Thanks
--
Gary
Gary,
A standard RJ11 telephone connector will work fine with the ports on
the back of the TDM card. I am assuming that in the UK, you use the same
connector as we do in the States.. :-)
Robert
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either the laptop's multimedia controller or 2000 seem to be f-ing up)
to test it, and I'll report the results to you when I do that.
Regards and thanks for your continuing support,
Robert Geller
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and
interference. Perhaps I ought to run out and buy a couple factory-made
cables to test the difference, if any, between them.
Regards,
Robert Geller
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Brian Capouch wrote:
Robert Geller wrote:
At this point, I'm thinking that it could be a matter of bad cabling
or something. The Cat5 cable that's running the 8 or so feet from my
PC to my router is homemade by me, and many people do report problems
with homemade cables. I may not have
Brian Capouch wrote:
Robert Geller wrote:
Wow, very interesting. Thank you so much! BTW, regarding YMMV, did
you have a separate, dedicated sound card? I don't -- it's integrated
into my motherboard. Would this still apply? Of course, there are
still ports in the back for in, out
Brian Capouch wrote:
Robert Geller wrote:
What should I be looking for in /proc/interrupts? If the first field
in each row is the IRQ, I don't see any of the same numbers listed,
so would that mean there are no conflicts?
Why don't you include the output in your mail?
B
it down? Should I permanently disable it and reboot?
Again, thank you very much for your ongoing help; I feel like I'm paying
(or ought to) for professional support here. :-)
Regards,
Robert Geller
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Robert Geller wrote:
Rich Adamson wrote:
Thank you very much for your response. I do acknowledge that my
previous posts did not contain much technical information to speak
of, but it was mainly because I wasn't/am not familiar with the
Asterisk CLI and troubleshooting Asterisk
Robert Geller wrote:
Robert Geller wrote:
Rich Adamson wrote:
Thank you very much for your response. I do acknowledge that my
previous posts did not contain much technical information to speak
of, but it was mainly because I wasn't/am not familiar with the
Asterisk CLI
= 2,1,Macro(rg-group,30,Operator,108-101-102) ; 30 sec wait, CID prefix
Operator:, rings x108,x101,x102
exten = 2,2,Macro(vm,108,1); goes to 108's voice mail if no-one
picks up
Maybe this post will help someone else.
--Rob
Quoting Robert G. Ristroph [EMAIL PROTECTED]:
Hi,
I set
be more flexible to do it from within asterisk.
Do I have to write my own agi application and call it from the same place
VoiceMailMain is called to do this ?
Thanks in advance,
--Rob
--
Robert G. Ristroph
Airlink Systems
[EMAIL PROTECTED]
(512) 231-1240 x103
want to figure out why I hear
scratchiness, skipping, and general lack of clarity on the other side.
*Please note that I can call in to my PSTN number on the Asterisk system
and hear the demo (Allison) pretty much perfectly, so it's definitely
not an Asterisk problem!*
Regards,
Robert Geller
dialparties is used.
Thanks in advance,
Rob
--
Robert G. Ristroph
Airlink Systems
[EMAIL PROTECTED]
(512) 231-1240 x103
This message was sent using IMP, the Internet Messaging Program
!
Regards,
Robert Geller
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conversation?
Thanks a lot, all.
Robert Geller wrote:
Hello all,
I am using a headset and the X-lite softphone (sometimes I use
IAXComm, but I'm having difficulties using OSS emulation with it) to
connect via uLaw to my internal Asterisk server, which is a Pentium II
400 with 128 megs of RAM
If nic is loaded using modprobe - you can set options for duplex -
depending on the nic...
See /etc/modules.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Monday, August 29, 2005 11:13 AM
To: Asterisk Users Mailing
it
worked.
SNIP
You claim it is an Asterisk issue, did you by any chance
make sure that database was allowing connections on
127.0.0.1 and localhost and not just the actual IP??
Robert
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U joke - duh!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Innocent Evil
Sent: Wednesday, August 24, 2005 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI + Ruby
What IDE are you
.
John Novack
P.S. Robert- Something wrong with your mail clock?
You responded to a message hours before it was sent!
Sorry, was not trying to insult your expertise.. Just
sometimes things can get overlooked.
Thanks for the heads up.. Something went awry with my
email server today and the ntp
Y'see it? There it goes! Right over his head.
Huddleston, Robert wrote:
U joke - duh!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Innocent
Evil
Sent: Wednesday, August 24, 2005 3:53 PM
To: Asterisk Users Mailing List
=192.168.77.254
dtmfmode=info
disallow=all
allow=g729
nat=no
canreinvite=yes
qualify=yes
Maybe double and triple check that the router context is
actually being used. SOunds like it isn't. I have gotten
caught in this situation before.
Robert
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I did something like this I had to work on the perl to get it working
myself.
I found if I put a wait and a beep in the dial plain for the calling user
then
they would get beep 3 or 4 seconds later this give the calls time to set up.
Worked great for me.
-Original Message-
From: [EMAIL
I say start small and then go big... Oh I don't know a Proliant 1500 or
3000 should work nicely -- if you can handle the noise =)~
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Latham
Sent: Monday, August 22, 2005 1:40 PM
To: Asterisk
and it works fine now, and
we haven't noticed a severe degradation in sound quality - most of my
operators were just happy the echo was gone :)
+1 here too:
Uncommenting AGGRESSIVE_SUPPRESSOR and recompiling took care of 99%
of my TE110P/PRI echo.
-Rob.
--
Robert Goodyear
Brand Up LLC
http
On Aug 18, 2005, at 3:07 AM, Stephen wrote:
Hi All,
How can I lock the extension in Asterisk?
For example , my extension is 1000 and I am away for business trip.
I want to lock my extension during my absence.
Can it be done in Asterisk?
regards,
Stephen
You could write a little script to
If you are using a TDM card its also important to use the new tool fxotune.
This should help as it will match the fxo card to the line. Hybrid balance
will help echo as well and I assume fxotune is helping to balance the line.
With a matched hybrids on both ends of a 2 wire interface you will
On Aug 12, 2005, at 7:06 PM, Tim Connolly wrote: I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or … any other brand and model that is known to work well with the TE411P ? Will an old Proliant
this.
(There is not a question in this message. Its
just meant to be informative.)
- Robert
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Does anyone know of a way to make a standard analog phone
plugged into an FXS port do something other than get a dialtone when you pick
it up? For example, if the phone should automatically ring someone or
play a greeting when picked up without having to enter an extension?
- Robert
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
r: Generate a ringing tone for the calling party, passing no audio
from
the called channel(s) until one answers. Use with care and don't
insert
this by default into all your dial statements as you are killing call
--
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
On Aug 4, 2005, at 2:10 PM, Robbie Hughes wrote:
I have a 12 channel PRI with SNOM 190's and asterisk CVS from January.
Most calls are fine, all incoming calls are fine, but I am getting
echo on a significant number of outgoing calls
/Nat - Softphone/hardphone(Location B)A great guide is here:http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.htmlPay very close attention to the externip and localnet parameters that belong in the GENERAL section of SIP.conf-- Robert GoodyearBrand Up LLChttp://www.brand
Has anyone tried this? I got in to download but now I can not get back
into mozdev.org. It did not come with any directions or help. If anyone
has it working where did you get instructions?
TIA
Bob
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Jean-Denis Girard wrote:
The project home page is:
http://moziax.mozdev.org/
(unfortunately mozphone.mozdev.org had already been registered but
nothing there).
If you have specific question, go ahead I'll try to help as much as I can.
I'm also very interested in feedback.
Thanks,
Thanks.
Robert A. Rawlinson wrote:
Thanks. I had just found it. I am trying to get it installed now. I will
let you know how it goes.
Bob
Install was easy. Now if I can just get [EMAIL PROTECTED] running. I will let you
know more when I get * up and running.
Bob
th getting ALERT_INFO messages with the 480i working.
Robert
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Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday...
rhuddleston.vcf
Description: Binary data
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On Thursday 28 July 2005 18:28, snacktime wrote:
On 7/28/05, wassim darwish [EMAIL PROTECTED]
wrote:
what is the most stable linux that we can build
business on it, i mean the best linux a linux without
problems .
I have Suse 9.1. I had no problems installing it. It is not
the latest
Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO.
I've been able to work the FXO ports out and been able to make and
receive calls using softtel PC phones. I'm having difficulty with
configuring 4 line non-PBX analogs to function on the FXS side tho..
I've
Er, make that TDM400P cards... X.X
rc
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]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert Chapin
Sent: Monday, August 01, 2005 9:13 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] with Analog Phones
Working [EMAIL PROTECTED] 1.3 two 4 port TDM100 WildCards, 3 port FXS, 4 port FXO.
I've been able
I saw on here where there was an asteriskathome site where I
could sign up for the mailing list. However when I bring up
that site I just get a blank page. Is
www.asteriskathome.org the correct address?
Bob
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Thanks to Doug Logan and The person at [EMAIL PROTECTED]
Bob
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On Wed, 27 Jul 2005 18:07:23 +0200
Walid Azab [EMAIL PROTECTED] wrote:
Hi..
I am trying to do something but it is giving me some
hard time here. I have
an IAX2 trunk to FWD which is registered and working
just fine. I have =
011|. as my dial pattern to allow that. But if I want to
dial a
I was installing my first [EMAIL PROTECTED] and every thing seemed to go ok
untill it got to a line that says:
find: /var/www/html/admin/: No such file or directory
find: /var/www/html/admin/: No such file or directory
There it still sits but not doing anything for over an hour now.
Has anyone
immediately. So what you have given him should work fine.
Robert
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for it.
Any help at all is greatly appreciated.
Thank you.
- Robert
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with the TDM cards?
- Robert
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On Mon, 25 Jul 2005 15:44:07 +0200
Alexis F. [EMAIL PROTECTED] wrote:
Hi,
I would like to use a digum card to call an external
number through my PSTN. I think that I have a problem in
the configuration. Asterisk returns me app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
Is anyone else having problems with nufone's inbound? When I try calling all I
get is either a fast busy signal or the recording that the person I'm calling
isn't available.
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On 7/26/05, Arnd Vehling [EMAIL PROTECTED] wrote:
Hi,
i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07
and everything worked fine sofar when suddenly the voicemail and
musiconhold sound output stopped working.
[snip]
I am using an installation with several
Hi All, I am just looking at using Asterisk now and the first thing I need to do is via pass two external numbers to asterisk and call out connecting the calls togther. These will be through our physical PBX connected to the asterisk server. We are essentially trying to connect two external
Hi,
I spot weird behaviour of latest Firefly 3rd party on my laptop. Sometimes
it comes to state that it won't start (hangs on Initializing ) and it
again works after system restart... Didn't yet figured out how to recreate
it.
Any similar experience ?
Also - how can I force Firefly to
On Wed, 20 Jul 2005 18:00:24 +0200
Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I spot weird behaviour of latest Firefly 3rd party on my
laptop. Sometimes it comes to state that it won't start
(hangs on Initializing ) and it again works after
system restart... Didn't yet figured out how
of ALL things Voip.
Flame away all...
Robert
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Thanks alot guys,
I will look further into the flash hook transfer. Essentially you are right we know we will incurr the cost of the calls . We are providing a marketing-consumer service to our clients, whom bill for the calls. So I need to investigate monitoring and reporting of the calls.
On Jul 18, 2005, at 3:13 PM, Michael D Schelin wrote:
Here is a letter I sent them for my $150 paper weight.
The forum is not a place to post ransom notes. You've added zero
benefit to any reader here, nor to yourself, since you didn't actually
ask a question in your email.
Hi All,I am just looking at using Asterisk now and the first thing I need to do is via pass two external numbers to asterisk and call out connecting the calls togther.These will be through our physical PBX connected to the asterisk
server. We are essentially trying to connect two external numbers
On Jul 20, 2005, at 12:22 AM, Brian Capouch wrote:
Michael D Schelin wrote:
Real scary who
You certainly have found an unusual way to promote your business.
B.
Kinda sounds like a schoolyard taunt, usually found near most lemonade
stands, doesn't it?
exten 301
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Robert
Is is possible to specify the MOH Class when defining a MeetMe
extension?
I tried
exten = 300,1,MeetMe(300|M(class))
But that did not work.
Thx,
-Rob.
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On Jul 14, 2005, at 11:17 AM, Robert Goodyear wrote:
Is is possible to specify the MOH Class when defining a MeetMe
extension?
I tried
exten = 300,1,MeetMe(300|M(class))
Replying to my own query, just in case anyone else is as dense as I
am...
exten = 300,1,SetMusicOnHold(confclass
Anyone know how to bypass the CONFIRMATION of the user announcement
recording in MeetMe?
While I like the please say your name to announce a user into a
conference, I find it confusing and time consuming to make the user to
press 1 to accept a recording they haven't even previewed.
I'm not
.
I will need assistance in planning and deciding about feasability and
also later in programming, deploying and supporting it.
I will prefer someone in germany, best near Hannover (my site) or
Chemnitz (customer's site).
The job should surely be paid for.
best regards,
Robert
, but this is a hardware-neutral
question.
Thanks.
--
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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Hi,
I'm aware that incoming and outgoing calls are going fine when isdn channels
are involved - caller id properly identifies calling party, so user can call
back
But how to properly handle this for iax, sip calls
I have few questions :
- BTW, what to type for instance in remote
Hi,
I'm using AMP and its dialparties.agi as most important script in system.
I'd like to port configuration to more embedded system, where I don't have
Perl available.
So I'd like to implement dialparties.agi functionality as closest as
possible with dialplan language.
Are there any
Hi,
I'm not sure if DTMF is convenient solution for user that has cellular on
his ear
Regards,
Rob.
- Original Message -
From: Dean Collins [EMAIL PROTECTED]
To: Ed Greenberg [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
- Original Message -
From: Richard Koch [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, July 08, 2005 4:38 PM
Subject: [Asterisk-Users] Speech Recognition
Ed,
Check this out:
http://turnkey-solution.com/asterisk-sphinx.html
That got me up in running in no
On Jul 8, 2005, at 12:43 AM, Jay Milk wrote:
All,
I'm currently only setting CID as a ten-digit number. Has anyone on
this list tested caller-id delivery with various services? Is there
*one* usable format (i.e. 1+10, or +1+10), or does it vary from
provider to provider?
Jay, FWIW the US
Is anyone having issues with audio being passed inbound via Teliax?
Trying to isolate an issue here.
Thx,
-Rob.
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At 15:21 06/07/2005 +0200, Tobias Wolf wrote:
Hi,
i was successful in compiling app_conference and setting up an conference
was quite easy. :-)
Does anyone knows if it is possible to have an IVR accessable from inside
the conference. So, if i dialed into an conference i want to be able to
At 15:31 07/07/2005 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Jean-Hugues ROBERT [EMAIL PROTECTED] wrote:
But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
that unfortunately it does not work for SIP channels due to the mixing
not being done in the zaptel
On Thu, 7 Jul 2005 10:49:32 -0700
Dan Adams [EMAIL PROTECTED] wrote:
Hi, I am sorta a newbie to the asterisk community at
least in the realm of
hardware types. I was wondering, what type of card is
used to allow asterisk,
on a slackware installation to talk to a standard phone
line so that
On Jul 4, 2005, at 2:43 PM, Jimmy Smith wrote:
you guys are so friggin funny..
We try. Meanwhile, you are SO illiterate; are you trying?
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and it states the number has been
disconnected.
Robert
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On Tue, 5 Jul 2005 22:07:11 +0800
Ian Bert Tusil [EMAIL PROTECTED] wrote:
I've just Installed [EMAIL PROTECTED] i browsed it's
built-in AMP. it
prompts for a login if you click on asterisk management
portal. i
tried
user:[EMAIL PROTECTED]
pass:password
and
user:admin
pass:password
but
I've got a handful of ATAs Innomedia that support two ports... I have one
plugged in for voice for the house and the other I use for dialup internet..
ONLY for testing newly built dial-up computers that they can get online and
surf... Gotten some pretty good speeds out of them too
Robert
On Tue, 05 Jul 2005 11:26:39 -0700
Bruce Ferrell [EMAIL PROTECTED] wrote:
I've gotten word from their Marketing VP. They are
doing some kind of massive move and expect to be down
until Thursday
Sounds like their Marketing VP needs to get a clue and let
customers know what is going on.
and recompiled and all
was well.
--
Robert Goodyear
Brand Up LLC
http://www.brand-up.com
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On Jul 4, 2005, at 12:17 AM, Bryce Chidester wrote:
Just a thought, but I seem to recall that in the dialplan, inlcude and
other similar statements are not prefixed by the hash character (#).
Try include = .
-Bryce
You're thinking of contextual includes, not filesystem includes --
which
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server.
On Jul 2, 2005, at 1:00 PM, Roland Zagler wrote:
sorry for the misunderstanding, robert!
the point is: during the caller is listening to the soundfile played to
him
the dialplan should continue to dial the sip phone 100 and after sip
phone
100 is answered and the announcement file is played
On Jul 2, 2005, at 8:33 PM, Ronald Wiplinger wrote:
Robert Goodyear wrote:
On Jul 1, 2005, at 1:47 AM, Ronald_Wiplinger wrote:
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I try
to dial an extension (from X-Lite), next
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Keith Caldwell
Sent: Saturday, July 02, 2005 8:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: TDM11B Dev Kit PCI + Asterisk CVS Head
I am confused about one of my installed server
The dial plan seems to be ok, but sometimes NOTHING happens if I
try to dial an extension (from X-Lite), next time it is done.
X-Lite does not have a tone, nothing and does also have no time
out. It seems it is not connected to the server.
of what you're after.
Perhaps you need to create a call file that then joins the two legs of
the call afterwards?
yes, robert, but how do i join the two legs inside a call file or
in the dialplan?
i have experienced that call files can do a call to a channel and
if this call is answered it can
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Sent: Saturday, July 02, 2005 10:49 AM
To: andrew matthews; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] passing through MWI info
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Robert Goodyear
Brand Up LLC
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On Jul 2, 2005, at 12:55 PM, Mahmoud Badran wrote:
try this one
exten = 999,1,Answer()
exten = 999,2,playback(~.mp3)
exten = 999,3,dial (sip/100)
exten = 999,4,playbackground(~.mp3)
exten = 999,h,Hangup()
not sure abt playbackground should be before the dial command or after
Mahmoud:
On Jul 2, 2005, at 12:19 PM, Roland Zagler wrote:
Thank you, Robert!
The announcementfile plays well, but at Dial-option m i have to
specify a MoH class,
that is something i cannot use (as i wrote in my post).
Noted, which is why I offered option two.
Background command waits for a user
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