How do I assign the MWI to a SIP phone on my asterisk server that is coming
from an ITSP?
I see the SIP message come across as having a message waiting but how does
one get that
to go to an extension on my box.
Thanks
Tom
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I am trying to use the Gotoiftime CMD to not allow calls to be placed
between the hours of 12am-5am, except if you know the PIN number to dial out
and if the call is for 911.
What is the best way to implement this solutions?
I have the gotoiftime like so:
exten =
yes, you can use Trixbox.On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote:
For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do
for the key presses that I put in but it will not stream the file back to me using Cepstral.Asterisk 1.2.9CentOS 4.2Thanks,Tom Vile
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Moore [EMAIL PROTECTED] wrote:
1) Try giving it an extension (say .gsm) and seeing if that works. Make
sure you change both the file and your script.
2) Does the rest of the script work? If you run './test.php', do you
get any errors?
Jay
Tom Vile wrote:
There are no file extensions
:
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*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Tom Vile
*Sent:* Wednesday, November 15, 2006 9:16 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Grandstream GXP2000
Just wondering if anyone is using the directory.agi app in AGI perl, I can't
seem to get the searching function working. It lists my contacts but when I
type in digits it say no match found.
Thanks
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On 11/14/06, Tim Uckun [EMAIL PROTECTED] wrote:
On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote: I need to write an app which takes a phone call, asks for the user to input a number and then queries a db via a webservice and reads the results a row
at a time back to the caller.First, is this
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I have a query that query's my database based on the read input for an ID number.exten = s,4,MYSQL(Query resultid ${connid} SELECT\ `FirstName`\ `HomePhone`\ FROM\ `contacts`\ WHERE\ `ContactID`= \'${ID}\')
exten = s,5,MYSQL(Fetch foundRow ${resultid} var1 var2) ; fetch rowProblem is is that when
I was wondering how the Directory CMD can read the input of numbers from a phone and translate that to search in voicemail.conf. Essentially I want to be able to look up contacts with MySQL and have the user input 3 digits corresponding to the contacts last name and have it search for it in the
?
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That probably because you are using Webmin. Just change the port Webmin listens on instead, I use 9000.On 11/6/06, Zeeshan Zakaria
[EMAIL PROTECTED] wrote:I'll keep that in mind for future. I read about using 10001 as start port on Nerd Vittles website.
Is there some good material online to
Yes, it can. I put all of my servers to 9000 though.On 11/7/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Webmin uses UDP?Audio is generally RTP over UDP.Tom Vile wrote: That probably because you are using Webmin.Just change the port Webmin
listens on instead, I use 9000. On 11/6/06,
He is not talking about Trixbox but FreePBX and his assumption is correct. Just load Asterisk and then FreePBX later.On 11/4/06, Zeeshan Zakaria
[EMAIL PROTECTED] wrote:No, you can't add this to asterisk install. Trixbox formats the drive and installs everything from scratch. But it installs all
If its always 2, then its waiting for CallerID or Fax Detection?On 11/4/06, Jordan Novak [EMAIL PROTECTED]
wrote:They are in Kewl start now but I have tried groundstart and
loopstart. Waht could i be missing that would cause this.I start with a
Exten= s,1,answer. I am using three FXS modules on
I love the Snom phones as well. The function keys are great and easy to use.On 10/31/06, mitcheloc [EMAIL PROTECTED]
wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons
can be retooled for any environment,
I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99.2 Asterisk boxes in different locations? Sure, you can do that and its quite easily.
On 11/1/06, Ken Williams [EMAIL PROTECTED]
it told you:Permission DeniedCheck the permissions on that directory.On 10/31/06, Zeeshan Zakaria
[EMAIL PROTECTED] wrote:Anybody knows why ARI gives this error message when I enter extension number and password.
Warning: file(/var/spool/asterisk/voicemail/default/222/INBOX/msg.txt): failed
Did you install PHP?On 10/31/06, Alok Mohapatra [EMAIL PROTECTED] wrote:
Hi All,
I
have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk
Management Portal (AMP) for web interface.
After installing properly when opening in the webpage it is
not parsing the index.php
http://www.digium.com/en/docs/misc/compatibility_notes.phpServer Compatibility The following list of servers are known
to be partially incompatible with Digium hardware. We do not recommend
using the following computers to set up an Asterisk server: Dell PowerEdge 1600Dell PowerEdge SC 420Dell
I used some of the ideas found here:http://www.voip-info.org/wiki/view/Asterisk+manager+ExamplesOn 10/30/06,
Steve Davies [EMAIL PROTECTED] wrote:
Hi,We are successfuly using TAPI with Asterisk in order to provide ageneric and fairly well supported interface from Windows desktops toAsterisk -
I was wondering if anyone has implemented a feature that would allow a user to record a phone call and once the call has ended, the call is forwarded to his voicemail?ThanksTom Vile
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On 10/23/06, mail-lists [EMAIL PROTECTED] wrote:
Hello,I've posted this at the trixbox and freepbx forums and haven't been ableto get an answer. I thought perhaps the guru's here might be able tohelp me out :)I'm having some issues with setting caller IDs. There are 2 problems
that I would like to
The o option is mentioned over at FreePBX and how to restore this setting.
On 10/23/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
mail-lists wrote: Hello, I've posted this at the trixbox and freepbx forums and haven't been able
to get an answer. I thought perhaps the guru's here might be
Is it an NTFS Share?On 10/19/06, Paul Gaffney [EMAIL PROTECTED] wrote:
Message: 12
Date: Tue, 17 Oct 2006
18:07:04 -0700 (PDT)
From: sdgesa gaeharth
[EMAIL PROTECTED]
Subject: Re:
[asterisk-users] Extremely choppy sound on some of
ourPOTSnetwork calls; goes away with
not having that issue with ChanSpy here but I loaded up a 1.2.12.1 box last night with a TE110P and Asterisk Crashes after receviing a call and I was using the latest zap drivers. I put in the Sangoma card and no problem. Must have been some motherboard compatibility.
On 10/18/06, Thomas Kenyon
I 4th it.On 10/18/06, Matthew Thompson [EMAIL PROTECTED] wrote:
On 17 Oct 2006, at 22:09, Richard wrote: I would have to second the Sangoma buy. Their tech support is second to none and more then helpful.
I've never had any problems with their products that wasn't my own fault.
Thirded - I've
seriously go look at voip-info.org for the answer, thats where we get most of our info from, or perhaps type show application chanspy from the asterisk CLI.Are we that lazy that we cant use google to search. Ridiculous.
On 10/18/06, Sergio R. D'Ippolito [EMAIL PROTECTED] wrote:
How
fine for me here since it came out. We are running 15 extension all day long.On 10/16/06, shadowym [EMAIL PROTECTED]
wrote:I am getting ready to image a production system.Right now I am planning on
using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3.I will be using aSangoma A200D card.I read of
http://www.polycom.com/products_services/1,1443,pw-17068,00.htmlOn 10/13/06,
Jessee J Holmes [EMAIL PROTECTED] wrote:
I've played with one, but they are not available yet from Polycom (as in shipping to distributors). They are only demoing the phones at the moment. I'm sure if anyone has gone to
315 extensions (1198 priorities) in 92 contexts.On 10/10/06, Jeremy McNamara [EMAIL PROTECTED] wrote:
One of the smaller systems:-= 9924 extensions (29772 priorities) in 6 contexts. =-Jeremy McNamara___--Bandwidth and Colocation provided by
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Thanks James, I was close.On 10/8/06, James Jones [EMAIL PROTECTED] wrote:
exten = _*1XX,1,Set(CALLERID(all) = Nursery ${EXTEN:1})exten = _*1XX,2,Dial(SIP/400)Tom Vile wrote: I have a need for a dialplan that call for the ability for people to dial *1XX and it send a call
to extension 400
I have a need for a dialplan that call for the ability for people to dial *1XX and it send a callto extension 400 with the calleridname of Nursery and the calleridnum of the *1XX number thatwas put in minus the *. Now I know how to do it individually but I now there must be an easier
way to simply
login as root and type help-aah and you will see a list of commands to change the admin password.On 9/30/06, Jim Lynch
[EMAIL PROTECTED] wrote:I'm looking for the username/password to access the web gui for freepbx
admin rather than the voicemail passwords.I need to reconfigure theextentions/ring
On 9/25/06, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
FYI, this is an asterisk mailing list, NOT trixbox. Most people on here don't care when trixbox is going to do something. Try their list.
On 9/25/06, Christopher Corn
[EMAIL PROTECTED] wrote:
I know asterisk 1.4 has t.38 pass through,
They only do numeric callerid.On 9/11/06, Ricardo Carvalho [EMAIL PROTECTED] wrote:
I have tested Grandstream Budgetone 102 and Grandstream Budgetone200 and with both, if they are called from a caller that is analphanumeric user, their display shows a unintelligible name impossible
to figure out
Teliax is not unlimited but has a cap of 2500 minutes per month.***
Softcap of 2500 Minutes (including 1000 minutes of toll-free inbound, if applicable).On 8/30/06, Crazy Boy
[EMAIL PROTECTED] wrote:Hi, Taliax has unlimited calling plan per month. You can see
WWW.TELIAX.COM Regards,
$24 per monthOn 8/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
What
cost do you pay per month for the 2500 minutes?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Tom
VileSent: 30 August 2006 13:54To: Asterisk Users Mailing
List -
Use the m option at the end of the dial string. Google told me so.On 8/26/06, Crazy Boy [EMAIL PROTECTED]
wrote: Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring
The Budgetone only supports a 12-digit caller ID LCDOn 8/15/06, Guus Houtzager [EMAIL PROTECTED] wrote:
On Tuesday 15 August 2006 17:53, Julian Lyndon-Smith wrote: Try without the
Set(CALLERID(all)=some text 123-123-1234Tried that, no effect, still shows only the number part, without the
Did it about 10x to friends. Pretty funny.On 8/10/06, Hugh L. Johnson [EMAIL PROTECTED] wrote:
http://snakesonaplane.varitalk.com
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Yes it supports PRIOn 8/9/06, Colin MacMillan [EMAIL PROTECTED] wrote:
Brian,What you need are some sources of good information to get you started. Based on what you wrote there is a lot to cover - impossible in an email.Buy and read the book from this link -
http://www.thetechguide.com/howto/asterisk/chanbluetooth.htmlOn 8/8/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Has anyone used a bluetooth phone (eg Cingular 8125 aka HTC Wizard) as anFXS/FXO with asterisk?I'd like to be able to route incoming cellphone calls into asterisk, aswell as make
Does your cingular 8125 have bluetooth? If so it should work. Mine works with Motorola and LG phones fine.On 8/8/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Have you used it with a cingular 8125?-DanOn Tue, 8 Aug 2006, Tom Vile wrote: http://www.thetechguide.com/howto/asterisk
Did you look on the site?http://www.4psa.com/products/voipnow/demo.phpOn 7/29/06, Dinesh Nair
[EMAIL PROTECTED] wrote:
On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think.As
powerful as
Turn off the call features in the phone, by default the *70 codes are enable in the phone so that the phone can do call waiting and such. If you want asterisk to do this you need to disable the feature codes in the phone.
On 7/28/06, Chris Bagnall [EMAIL PROTECTED] wrote:
I just bought a grand
does it reach the asterisk console? and have you turned off the dial features in the phone?On 7/28/06, Cavanna, Richard
[EMAIL PROTECTED] wrote:Here is the software version:Program--
1.1.0.16Bootloader-- 1.1.0.1When I pick up the line and dial *70 it just disappears and never dials.If I enable
/usr/src/asteriskOn 7/28/06, Wasif [EMAIL PROTECTED] wrote:
Hi,I am using TriBox 1.1.1/Asterisk. I want to know where I can find sourcedirectory of Asterisk in system so I can install Asterisk audio conversionmodule (
http://redice.krisk.org/res_conv-0.1.tgz) to convert ulaw promptsinto g729
you can trunk the two boxes together with IAX. Check out trixbox.org and search, its been covered a few times.On 7/21/06,
Gidean Chan [EMAIL PROTECTED] wrote:
Hi!
Does anyone know how to connect 2 AAH
IPPBXs so that one extension in A IPPBX can use the PSTN trunk in B IPPBX for
dial out?
http://www.voip-info.orghttp://asteriskguru.org/tutorials/On 7/20/06,
David R. [EMAIL PROTECTED] wrote:
Here's my situation. I'm a programmer at the ISP I work for and my supervisors have seen fit for me to become an administrator for our new ventures into the VoIP world. All of us are relatively
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Fax
For the NIC setup you can bond 2 cards together for redundency. Take
a look here for some more info on bonding.
http://www.redhat.com/docs/manuals/enterprise/RHEL-4-Manual/ref-guide/s1-networkscripts-interfaces.html#S2-NETWORKSCRIPTS-INTERFACES-CHAN
On 7/13/06, shadowym [EMAIL PROTECTED]
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will
2. It uses meetme
3. I need to be able to complete the xfer.
Thank You
On 7/11/06, Tom Vile [EMAIL PROTECTED] wrote:
Have you tried using the page() feature. It has a d option for full
duplex but if you use it without the d it will be one way..
* d - full duplex audio
* q
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the all-new Yahoo! Mail Beta.
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afficionados are both called users?
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]@sip.broadvoice.com in the asterisk
console being repeated every 20 or so seconds?
On 6/19/06, Tom Vile [EMAIL PROTECTED] wrote:
your context=incoming does not match your context in extensions.conf
incoming-bv
On 6/19/06, John Klimek [EMAIL PROTECTED] wrote:
Asterisk seems to register just
(in-system) or makes a long distance call. She just gets a fast busy,
and I see no output on the console whatsoever.
[EMAIL PROTECTED] wrote:
Message: 1
Date: Wed, 21 Jun 2006 13:44:58 -0400
From: Tom Vile [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing
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://www.io.com/~hcobb/
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://www.crompton.com *
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book. It precludes me from using any DTMF over PSTN with *
at this point.
Any further help or explanation would be appreciated.
On Tue, 6 Jun 2006, Tom Vile wrote:
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.
On 6/6/06, Doug Crompton [EMAIL PROTECTED
/spa-3000/* and listen on both while pressing dtmf do you hear good
clean tones of enough duration to allow detection, in both directions?
Do you access DTMF required services over pstn, like banking, vm, etc
from local * system?
Doug
On Tue, 6 Jun 2006, Tom Vile wrote:
Using AVT in my sipura
in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug
On Tue, 6 Jun 2006, Tom Vile wrote:
try setting dtmf playback length to .5 in the admin section of the
Sipura
in and '*' it to get PW prompt.
Doug
On Tue, 6 Jun 2006, Tom Vile wrote:
But access to voicemail in Asterisk will not work with inband.
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termination. In other words you can only receive calls
from the Voxbone network.
On 6/2/06, Tom Vile [EMAIL PROTECTED] wrote:
try voxbone.com
On 6/2/06, David K Parker [EMAIL PROTECTED] wrote:
Does anyone know of a good VOIP provider that I can obtain a DID from
Latvia? I live in the US
the best spam protection around
http://mail.yahoo.com
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Phone
http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+2#297UpdatingAsteriskmanuallypreferredsoyo
On 5/31/06, David K Parker [EMAIL PROTECTED] wrote:
hmm, I couldn't find any upgrade procedures there, only initial install
On 5/31/06, Tom Vile [EMAIL PROTECTED] wrote
/qD2uhr5AmyAfw=
=rfHQ
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Tom Vile
Baldwin
$2.50 p/month for 800 DID.
On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:
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They bill you for having the 800 number? I thought they only did that
for Michigan DIDs. They only bill my actual call time.
jens
On 23 May 2006, at 16:54, Tom Vile
were advertised as
free of monthly fees, call fees only.
jens
On 23 May 2006, at 20:13, Tom Vile wrote:
$2.50 p/month for 800 DID.
On 5/23/06, Jens Vagelpohl [EMAIL PROTECTED] wrote:
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They bill you for having the 800 number? I
/asterisk-users
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Asterisk-Users
of poer supply reduced noise to certain extend but still its
very clearly audible.
A good fix would be really helpful to me
Thanks
On 13/05/06, Ben Holt [EMAIL PROTECTED] wrote:
Darrick Hartman wrote:
Tom Vile wrote:
Same problem with audio quality. Got rid of them. Also the context
line
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Tom Vile
Baldwin Technology
:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, May 11, 2006 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000
What version of Asterisk?
On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:
I am
in the future.
On 5/12/06, Darrick Hartman [EMAIL PROTECTED] wrote:
Tom Vile wrote:
Same problem with audio quality. Got rid of them. Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12
line.
Also, if it is dialing each phone line by line will there be a delay for all
the phones are dialed and pick up.
Thanks again.
Forrest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Friday, May 12, 2006 10:21 AM
To: Asterisk Users
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
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