In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
says...
> When somebody calls me on fxo4 port * sents that call to SIP 214 phone.
> The problem is that when call ends and SIP user hangs up, the line stays
> up. Now I don't use "Congestion" any more. Can sombody tell me do I
> realy need that
On Tue, 27 Dec 2005 20:54:46 -0800 (PST), [EMAIL PROTECTED] wrote
> I'm looking for a 4-port external sip fxo which doesn't suck...
Michael Graves replied:
> How about a non-traditional solution...skips FXOs altogether. After
> much experimentation with SPA-3K, TDM400p and X101p I abandoned the
>
I've got a Grandstream GXP-2000 with FW 1.0.1.12. Using a "DanaCom HA 25"
earphone I have NEVER had any issues with the sound quality, local or
far-end.
A customer of mine has chosen to dump all of their GXP-2000's and instead
purchase Aastra 480i's, as the older generation had to turn the volume s
Pisac schrieb:
I'm reading voip-info... and it's only confusing me:
zaphfc, zapbri driver package, bristuff...
So, what to download and install? If I install bristuff from
junghanns.net, should I also install something else (patch)?
What is (and where is) that zapbri driver package?
Go to
hi all, I m a newbie to asterisk. I have just installed X1OOP card on my PC.In that PC i have installed asterisk. I have configured three files zaptal.conf,zapta.conf,extensions.conf for cheking X100P card's working ; I did following modifications in Zaptel.conf : fxsks=1 loadzone=u
Thanks, but I'm looking for information on porting numbers when the current
provider holding the numbers goes out of business and is unreachable. Can I
get the numbers? The business has had the same phone number for almost 30
years and definitely can't lose the number due to some provider's
insta
What are you using for AGI
The correct command to send
Would be:
EXEC Set(${CALLERID(num)}=0005551212)
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rehan AllahWala
> Sent: Thursday, December 29, 2005 7:01 PM
> To: C F
> Cc: asterisk-user
Look at:
http://bugs.digium.com/view.php?id=6077
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Robert La Ferla
> Sent: Thursday, December 29, 2005 10:43 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asteris
William M. Sandiford wrote:
I just upgraded my system to the latest svn-trunk
I previously made extensive use of the SetAccount() function, but now
I'm getting the following error
Dec 29 20:54:08 WARNING[4925]: pbx.c:1679 pbx_extension_helper: No
application 'SetAccount' for extension (voip
HI,
> Anyone have any info on porting numbers away from a VoIP provider to a Ma
> Bell or the like? Thanks!!
I had a friend port his from Bell ->VOIP ->VOIP. He had no trouble.
I would use a couple providers. So this way if one goes down there is a backup.
--
Leonard Burton, N9URK
[EMAIL P
Andrew Latham wrote:
I think most all of the phones have a hack to get it working. Aastra
analog ADSI phones even work as I read some where...
It works with the 9133i. This is a great feature!
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Alexander Lopez wrote:
I vote for 'a' as the auto-play option.
http://bugs.digium.com/view.php?id=6090
I second the vote. I thought of using the same letter after reading
your reply.
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Ast
I vote for 'a' as the auto-play option.
http://bugs.digium.com/view.php?id=6090
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> BJ Weschke
> Sent: Thursday, December 29, 2005 1:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
Is it possible to dial with a silent ring? If so, is it configurable
with * or does the phone have to support it?
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On Thu, 2005-12-29 at 16:22 -0300, Javier Ergas wrote:
> I have tried both inband and outofband too unsuccessfully. I think the
> priindication parameter says how Asterisk reports Busy and Congestion
> to the PSTN, not the other way around.
>
> In the Asterisk config sirrix.conf
> (http://www.voip
Thanks.
I do plan on keeping a POTS line for
fax/credit card machines and 911.
Anyone have any info on porting numbers away from a VoIP provider to a Ma Bell or
the like? Thanks!!
-Ross
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sen
I had heard the one about the Microsnot style strong arm tactic. Perhaps
we should lay this at the door of the MythBusters.
The really spooky thing is when one calls a company in the UK whom you
have never had dealings with before and hear your own voice talk to you.
This happened to me about
-- Personal opinion alert --
Do not route everything to an ITSP. At minimum keep a main
PSTN line with call forwarding or call forwarding on busy until you are 1%
confident that the service works, is reliable, stable, and will have some
staying power.
Kerry
GarrisonDirector of Techni
I just upgraded my
system to the latest svn-trunk
I previously made
extensive use of the SetAccount() function, but now I'm getting the following
error
Dec 29 20:54:08
WARNING[4925]: pbx.c:1679 pbx_extension_helper: No application 'SetAccount' for
extension (voipsubscriber-in, x
Hi,
I have been running Asterisk in Sweden for almoast 7 months now, with a TDM400P
FXO/FXS card.
Two things have been enoying though:
1. Asterisk did not hung up correctly
Solved with Asterisk 1.2.0 and patch from issue 0003874 (answer and
hungup on polarity switch Fix )
Than
I’m looking to move one of my clients to an Asterisk
system and a VoIP provider (Teliax, Voxee, ViaTalk, Voicepulse). My
concern is porting my client’s numbers to a VoIP provider. Let’s
say we get all their numbers ported to Teliax (or Voxee or viatalk, etc.),
everything is peachy for a y
On Thursday 29 December 2005 19:43, Wolfgang S. Rupprecht wrote:
> I wonder if these same phones with a decent in-the-ear earphone and a
> mini boom-microphone would have the same problems.
Of course not; the earphone and boom mic are far far far less mechanically
coupled. :-)
-A.
_
My clients that are using Plantronics headsets have virtually no complaints
with the GXP-2000's.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message--
Andrew Kohlsmith <[EMAIL PROTECTED]> writes:
> Honestly you said it yourself though... they are turning it up too
> high and pushing the audio beyond what its design specifications
> are. This is perhaps the fault of the software guys, as they allow
> you to go beyond what what the acoustic coupl
I'm reading voip-info... and it's only confusing
me:
zaphfc, zapbri driver package, bristuff...
So, what to download and install? If I install bristuff from
junghanns.net, should I also install something else (patch)?
What is (and where is) that zapbri driver
package?
Im using kernel 2.
And as an independent contractor she can pick and choose who she wants to do
work for. Big whoop dee doo.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, December 29, 2005 4:10 PM
To: Asterisk Users Mailing List - Non-Commerci
"Alison is a contractor and so whomever pays her money gets her voice."
Apparently not according to some people on this list she was
'unavailable' for another voip project about 6 months ago.
I don't remember/care about the details but that was the story.
Cheers,
Dean
> -Original Message-
Do u know how to instert it in the agi ?
$AGI->exec("SetCIDNum(8504338555)");
but it didn't work
> www.voip-info.org/wiki-asterisk
> or you could try the CLI show application Set, and show function
> CALLERID
>
>
> On 12/28/05, Rehan Ahmed <[EMAIL PROTECTED]> wrote:
> > Hi
> >
> > C
On Thu, 2005-12-29 at 16:43 -0500, Franklin Webb wrote:
> Greetings fellow list members,
>
> I am using ABE and I am attempting to impliment transfers using "#".
> I am using both "T" and "t" as options in my Dial() command. I am
> attempting to hit "#" then enter another extension from my dial
Yes they do use Asterisk for some of their facilities.
However, Alison is a contractor and so whomever pays her money gets her
voice.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Joe Pukepail wrote:
I heard on the radio about 1-800-FREE411 and tried it out, I was very
suprised to he
On 15:41, Thu 29 Dec 05, Elazar Rosenthal wrote:
> Has anyone gotten voicemail storage over odbc working with postgres? I have
> been trying to get this working and keep hitting snags.
Can someone tell me why ppl want to store the
voicemailmessages in a database ?
There is already a very good dat
Thanks for the reply, I'll give that a
try. Does anyone know why the zaptel drivers insert a 5secs pause before dialing
the last digit? there is a digium bug report about this, but they wrote it off
as they rekon they need the pause for echo training, sounds fishy to me...
anyone know how to
Go to ebay, search on x100P, there are always several for sale.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Murphy
Sent: Thursday, December 29, 2005 2:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users
I think most all of the phones have a hack to get it working. Aastra
analog ADSI phones even work as I read some where...
Not the phones, but Asterisk needs to have a 'hack' to get this working.
So, there must somethere i * code be a list of phones that has been
implemented.
PABX phones
Any idea what version of Asterisk ABE is based
on?
PaulH
- Original Message -
From:
Franklin Webb
To: asterisk-users@lists.digium.com
Sent: Friday, December 30, 2005 8:43
AM
Subject: [Asterisk-Users] transfers using
# in asterisk
Greetings fellow list
Just a quick one - did you do 'make samples' as part of installing asterisk?
That would have given you something to work with, at least.
(all of the files are in the configs folder in the Asterisk src folder if
you want to peruse them)
PaulH
- Original Message -
From: "Nitesh Divecha" <
I am using T1 Signaling and seeing the same problems (I think), so I don't think its just E1.
On 12/29/05, Javier Ergas <[EMAIL PROTECTED]> wrote:
I have tried both inband and outofband too unsuccessfully. I think the priindication parameter says how Asterisk reports Busy and Congestion to the P
I heard on the radio about 1-800-FREE411 and tried it out, I was very suprised to hear allisons' voice for the digits. Not sure if they are using asterisk for the backend on this or not.
Try it out its Free!
http://www.snopes.com/inboxer/nothing/free411.asp
(not afflicated with it in any way)
Hmm, did a search, didn't come up with anything under X100 or clone
X100, I assume you're talking about a few specific models, any ideas
which?
Thanks,
Matt
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> John Novack
> Sent: Thursday, Decemb
The line that reads:
exten => 6153247060,1,Wait(2)
should have been:
exten => 55,1,Wait(2)
Nitesh Divecha wrote:
Thanks James,
That should help to start my project Thanks a million...
I will keep on updating..
And thanks to all for the inputs
Thanks,
Neal
On Dec 29, 2005,
I have a very interesting project to put together in the southern California
area and am looking for anyone locally (preferably) that would be interested
in being involved in it. I cannot go into too many details but it does
require some good Asterisk configuration skills. Please email me off-list
Thanks James,
That should help to start my project Thanks a million...
I will keep on updating..
And thanks to all for the inputs
Thanks,
Neal
On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:
Nitesh Divecha wrote:
> Are there any examples of dial plans? Like how to make the defau
Use one of the clone X100 cards, available on eBay for 10 bucks or so
Not all PCI Voice modems work
Only ones with a certain chipset
John Novack
Matt Murphy wrote:
Hi all, I'm a freshfaced asterisk n00b, and I've got a dumb question.
(tm)
I'm experimenting with an asterisk at home install
Greetings fellow list members,
I am using ABE and I am attempting to impliment
transfers using "#". I am using both "T" and "t" as options in my Dial()
command. I am attempting to hit "#" then enter another extension from my
dialplan. I have tried this on both ends of the conversation a
Use Bristuff2005/12/29, Pisac <[EMAIL PROTECTED]>:
What is the easiest way to install and use HFC-S card on Asterisk?As less kernel compiling & driver installations as possible.Is it mISDN, or chan_capi, or vISDN, or zaphfc, or?___
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whats kernel version ? check in "dmesg" for system messagesCheers,Giovanni Miano2005/12/29, Dushyanth Harinath <[EMAIL PROTECTED]
>:Hey guys,Asterisk Server Specs :Asterisk version :CLI> show version
Asterisk SVN-trunk-r7230 built by [EMAIL PROTECTED] on a i686 running Linuxon 2005-12-25 16:14:47 U
On Thu, Dec 29, 2005 at 12:51:47PM +0100, Olle E Johansson wrote:
> I usually do
>
> asterisk -rvn | tee /tmp/sipdebug.txt
>
> Then turn on sip debug on the cli. This captures everything.
> You need to make sure that the debug output is sent to the console in
> logger.conf
script
Try to append # or * to numberDial(ZAP/g0/0199255#) or Dial(ZAP/g0/0199255*)Cheers,Giovanni Miano
2005/12/29, Eck <[EMAIL PROTECTED]>:Hi,
Sorry if this is a little off topic as its really more zaptel related, but hopefully someone will have come across this.,I am noticing a 4-5 second paus
I have a sample of the Yoda VG400 and I am having a devil of a time
trying to get all four channels to register to Asterisk. I have an
Asterisk 1.2.1 server.
I have tried adding one at a time and rebooting it, but it stops after
the first.
http://www.yoda.com.tw/model.php?type=Enterprise_VoIP&
Has anyone gotten voicemail storage over odbc working with postgres? I have
been trying to get this working and keep hitting snags.
Elazar
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Ok,
That is the place where I download the procedure, but I didn't found
anything about editing the Makefiles.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio
Sent: Thursday, December 29, 2005 3:16 PM
To: Asterisk Users Mailing L
Hi all, I'm a
freshfaced asterisk n00b, and I've got a dumb question. (tm)
I'm experimenting
with an asterisk at home install on a spare machine here. It has a PCI modem
installed in it. Zapatel seems to have recognized this and configured trunk
ZAP/g0. It does not, however, seem to wo
If you check the AsteriskGuru.com tutorial about this, he explains how to
edit this files manually.. it is really simple!
- Original Message -
From: "Carlos Alperin" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, December 29, 2005 5:59
Try visiting CES next week, it might be announced there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hunt, Bill
Sent: Thursday, December 29, 2005 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Linksys S
On Friday 30 December 2005 07:19, Blake Krone wrote:
> > Hey everyone I have my Asterisk server setup as the DMZ on my Linksys
> > router. If I use the internal IP as the domain in Xlite clients will
> > register and work, however, if I use the FQDN for my asterisk server the
> > clients will not r
Not out, nor expected in the near term.
Cory Andrews
Purchasing Manager
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct - 716.250.3402
mobile - 716.907.4054
email - [EMAIL PROTECTED]
AIM - b2Cory
- Original Messag
Do I need to compile first the app_rxfax.c & app_txfax.c to get the .so
files? If the answer is yes, how I do that command, just I'm not and expert
on GCC.
Thanks,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pierre Burton
Sent: Thursd
Anybody know the status of the Linksys SPA-942? Is it out yet?
-Bill
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Make it manually, because there is somme diff from 1.0.9
edit Makefile and add :
everything after +
Pierre
Carlos Alperin wrote:
Ok,
Everything was fine up to the moment to run patch < apps_makefile.patch
Then I got Hunk 1 of 2, on the line 98 of the Makefile.
This is the Makef
Can this work with any ADSI phone?
Can you send some links. The documentation is quite hard to find..
Thanks
Jacques
Andrew Latham wrote:
I think most all of the phones have a hack to get it working. Aastra
analog ADSI phones even work as I read some where...
On 12/29/05, Robert La Fer
Ok,
Everything was fine up to the moment to
run patch < apps_makefile.patch
Then I got Hunk 1 of 2, on the line 98 of
the Makefile.
This is the Makefile.rej output. As you
can see, the line 98 includes some + signs that are in the apps_makefile.patch.
[EMAIL PROTECTED] apps]
The
word from Kevin Fleming and Digium is that the use of realtime to support
multiple Asterisk boxes sharing sip is not supported or even known to work at
this point.
-Original Message-From: Asterisk
[mailto:[EMAIL PROTECTED]Sent: Thursday, December 29, 2005 12:14
PMTo: aste
I have tried both inband
and outofband too unsuccessfully. I think the priindication parameter says how
Asterisk reports Busy and Congestion to the PSTN, not the other way around.
In the Asterisk config
sirrix.conf (http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf)
I think most all of the phones have a hack to get it working. Aastra
analog ADSI phones even work as I read some where...
On 12/29/05, Robert La Ferla <[EMAIL PROTECTED]> wrote:
> So I can set it up to call a bunch of extensions and broadcast a message
> to them without the user picking up? Ca
1GB gratis, Antivirus y Antispam
Correo Yahoo!, el mejor correo web del mundo
Abrí tu cuenta aquí___
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I am
working an a multiple box asterisk solution. I need phones to be able to login
to multiple asterisk servers. I
need Phone A to be able to register to switch A and call Phone B that is
registered to switch B.
With
rtcachfriends=no this can be done,
However I then loss MWI and “sip
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The toneduration setting seems to have fixed it. Thanks for the tip!
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, December 29, 2005 7:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Aster
Has anyone used a Gnet VP168S with Asterisk? I've been testing with softphones,
and this will be my first attempt at using a hardware product to connect a
standard POTS telephone. The limited specs I found online suggest it should
work (SIP one FXS port and one PSTN "Fall back" port, but like I sai
Hi all,
I am installing an Asterisk box equipped with the Sangoma A102 card. The telco
just tested the PRI interface and it is ll ok. I
now connect my Asterisk box and I can't get the D-Channel up. If I enable
intense pri debug I see messages like the following:
--SNIP START--
< [ 02 01 7f ]
<
On 12/29/05, Alexander Lopez <[EMAIL PROTECTED]> wrote:
> I believe that there currently is no option for "Auto-play"
>
> You would have to edit the source code for that.
>
That is correct. But, I think it's a good idea, so be looking for a
bug on Mantis shortly to provide a new option for it. :
So I can set it up to call a bunch of extensions and broadcast a message
to them without the user picking up? Can I do this with Aastra phones?
This would be great for announcing incoming calls. "You have a call
from XXX . Press 1 to pickup Press 2 to send them to voicemail."
___
I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of "Do-De-Dah The number of have reached is not in service ". PRI Debug below.
-- Executing
www.voip-info.org/wiki-asterisk
or you could try the CLI show application Set, and show function CALLERID
On 12/28/05, Rehan Ahmed <[EMAIL PROTECTED]> wrote:
> Hi
>
> Can you send any example of this command like
>
> Set(CALLERID(num)=value)
>
> Thanks
>
> Rehan
>
>
> On 12/28/05, C F <[EMAIL PRO
Anyway around that? It's a PITA to have to change that all the time with my PDA & laptop.On 12/29/05, Kerry Garrison <
[EMAIL PROTECTED]> wrote:
If the machines with X-Lite are on the local network, use
the private ip, if they are outside the network, use the public
ip.
-Kerry
From: [EMAIL
Hi all,
I have a couple of LD PRI through Broadwing. I'm trying to verify that
I get the correct cause codes during the hangup. Specifically, I want
to know when a number is disconnected. All of the numbers I have tried
give cause 16. I have gotten a number to give cause 31.
Does someone
If the machines with X-Lite are on the local network, use
the private ip, if they are outside the network, use the public
ip.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Blake
KroneSent: Thursday, December 29, 2005 9:49 AMTo: Asterisk
Users Mailing List - Non-Com
Want to page all the SNOM phones in the office? Create a second SIP
account set to auto answer.
OFFICE=SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506&SIP/507
[default]
; Paging - Office only
exten => 44,1,NoOp(Paging the office)
exten => 44,n,SIPAddHeader,Call-Info: sip:192.168.20.1/; anwser-af
Hi
I am running asterisk SIP on port 5060, in my
sipura i changed the 5060 port to 6060. but it's still tring to register it to
asterisk.
how come this is possible,
Regards
Kani
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Asterisk-U
Paging is a quite common feature on modern PABX's and means that anyone
can connect to any speakerphone to broadcast messages, in some cases
even if the phone is in use. The typical usage would be the company
secretary desperatly trying to get hold of someone and since that person
don't answer
Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified
Yes.
[EMAIL PROTECTED] wrote:
Hello All,
Have anybody test ISP BILLING SYSTEM ?
http://ibs.sourceforge.net/index.html
Regards
Harry
___
Nouveau : téléphonez moins cher avec Yahoo! Me
we have some problems with asterisk coredumping.
We are running 1.0.9 on an Linux Debian Sarge with 2.4.31 Kernel.
Inside is an wct4xxp (4 E1s).
We terminate: SIP => Asterisk => DSS1
(gdb) bt
#0 0x4052c831 in q921_transmit_iframe (pri=0x401e0938, buf=0xbe5ff444,
len=9, cr=1) at q921.c:384
#1 0
In a few days it will be publicly announced.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Augustyn
Sent: Thursday, December 29, 2005 8:26 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Linksys SPA-
Where can I get more info on this product?
Thanks
robert
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Cory Andrews
> Sent: Thursday, December 29, 2005 1:23 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Aste
Why would you use this? Can someone please elaborate on the below
description? I'm missing the intent of it.
localhost*CLI> show application Page
[Synopsis]
Pages phones
[Description]
Page(Technology/Resource&Technology2/Resource2[|options])
Places outbound calls to the given technology / re
Hi,
Sorry if this is a little off topic as its really more zaptel related, but
hopefully someone will have come across this.,
I am noticing a 4-5 second pause when my Digium TDM21B is dialing, just before
dialing the last digit.
This is causing me problems here in the UK as some telco (no pri
Hey guys,
Asterisk Server Specs :
Asterisk version :
CLI> show version
Asterisk SVN-trunk-r7230 built by [EMAIL PROTECTED] on a i686 running Linux
on 2005-12-25 16:14:47 UTC
System details :
Centos 4.2 (Final)
Linux ip-pbx 2.6.9-22.ELsmp #1 SMP
Intel Dual Xeon 3.06Ghz
Intel SE7501CW2 Motherboa
I believe that there currently is no option for "Auto-play"
You would have to edit the source code for that.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Robert La Ferla
> Sent: Thursday, December 29, 2005 10:49 AM
> To: [EMAIL PROTECTED];
Tomislav Parcina wrote:
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED]
says...
I want to create an extension that goes directly to my new messages
without having to press "1". How do I do that? I can call
VoiceMailMain but then I have to choose "1" from the menu. I'd like it
to go the
Try adding a w in your dial statement. Asterisk will dial even if the
line is not ready with a dialtone, adding a w will wait a bit and then
dial the number.
On 12/29/05, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> Thanks, I will try that.
> -Kerry
>
>
> -Original Message-
> From: [EMAIL
On 29 Dec 2005, at 15:28, chris songer wrote:
has anyone had any luck compiling and installing the smsq.c
utility. I went through the tutorial online and found i was getting
errors all the way through it.
this is the tutorial i was using...
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
any
You REALLY don't want to have call waiting on a line going into any PBX. You
are only asking for problems. My basic home setup is an SPA-3000 but the
PSTN line only has call forward on busy, when busy, the number is forwarded
to a DID at iax.cc.
Kerry Garrison
Publisher - http://GeekGazette.com -
See the message I post right before this one for a simple example.
Ray Yang wrote:
Apart from the dial plan issue, can anyone let Asterisk act like Cisco GW to
accept SIP call without registered in advance?
I've tried this for a long time but no answer yet.
_
Nitesh Divecha wrote:
> Are there any examples of dial plans? Like how to make the default
> context?
>
> I just need a kick start on the config part, as I am really struggling
> on routing the calls.
>
Here is a very very simple example using a PRI. You will need more error
routing in a real
What is the easiest way to install and use HFC-S card on Asterisk?
As less kernel compiling & driver installations as possible.
Is it mISDN, or chan_capi, or vISDN, or zaphfc, or?
___
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Asterisk-
He had run into a deadlock situation where he entered an (illegal)
string for the dial plan that made the phone lock up right after reboot.
That bug was fixed in one of the early 4.x versions. The way out was a
little trick with the web browser.
Generally I think if people have a problem today the
has anyone had any luck compiling and installing the smsq.c utility. I
went through the tutorial online and found i was getting errors all the
way through it.
this is the tutorial i was using...
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
any light on this subject would be greatly appreciated
Hi,
I've got my * machine running, and it's connected to the pstn via a
Sipura SPA-3000. My PSTN line has the call waiting feature and I was
wondering how * deals with that. All incoming calls are prompted to
enter the desired extension, so I was wondering what happens when I'm on
the phon
On Friday 23 December 2005 00:39, Steven Ringwald wrote:
> On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote:
> > Try loading
> > http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if
> > that was in the line 1) while the phone boots up (keep your finger on
> > the reload
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