I think I found the part of my AGI the script is stuck at.
The #!/usr/bin/php command was fine. What the agi debug I believe is
displaying is the output of this:
$in = fopen("php://stdin","r");
Which explains what I thought was "cached"
-- the same #!/usr/bin/php5 -q command repeatedly failing
Thanks...completely irrelevant in my case though. I installed and set up
php on my end, so I have short_open_tag to be on :)
Any other ideas anyone?
Leah Newmark
VoIP Programmer
Capalon Communications
_
Probably unrelated, but
The script runs fine command line.
I have edited in the past to try as /usr/bin/php -q and it didn't help.
Right now, it's not even reading the changes. I must be missing
something very obvious...
LN
asterisk-users-requ...@lists.digium.com wrote:
> Message: 16
> Date: Wed, 24 Jun 2009 17:17:59 -0
Take a look at this:
/var/lib/asterisk/agi-bin/olehphone# head incoming.php displays:
#!/usr/bin/php
Message: 18
> Date: Wed, 24 Jun 2009 16:23:17 -0500
> From: "Danny Nicholas"
> Subject: Re: [asterisk-users] PHP AGI Not Working and Odd Behavior
> To: "'Asterisk Users Mailing List - Non-Commerci
Thanks for the suggestion, but I'm editing directly on the server
I've been doing AGIs for, what, 4 years now? I have never been *this* stumped!
__
On Wed, 24 Jun 2009 15:43:14 -0400, Leah Newmark
http://lists.digium.com/mailma
Thanks.
I didn't change anything in my dialplan. I am aware of reloading configuration
:)
My AGIs are copied from a working asterisk install -- the shebang argument is
how I've always done it. Either way, I have tried it without the -q as well,
and that also didn't succeed.
I just tried your t
#x27;s not in some other directory.
Any input: obvious or not is requested...a few people here are stumped!
Thank you!
Leah Newmark
VoIP Programmer
Capalon Communications
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asteri
esher
<[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Blank
Voicemail.Conf after Password Change To: "Asterisk Users Mailing List -
Non-Commercial Discussion" Message-ID:
<[EMAIL PROTECTED]> Content-Type:
text/plain; charset="iso-8859-1" On Monday 03 November 2008 1
Re: [asterisk-users] Blank Voicemail.Conf after
Password Change To: Asterisk Users Mailing List - Non-Commercial
Discussion Message-ID:
<[EMAIL PROTECTED]> Content-Type: text/plain;
charset=us-ascii On Wed, Oct 29, 2008 at 01:13:56PM -0400, Leah Newmark
wrote:
> > From time to t
rom voicemail is showing up in the asterisk logs, nor
does the console show any error after changing a password.
Any assistance on this strange behavior is much appreciated!
Thank you,
Leah Newmark
VoIP Programmer
Capalon Communications
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-- Bandwi
tarted calling on
his Verizon line, and
that sometimes has problems as well. I think we see this behavior most with
him, since he's a big client
and leaves lots of messages. There have definitely been other clients calling
that we don't hear the message
from.
I have not noticed any patt
read exactly the same in binary...but no
sound data is actually in there.
Is this a bug that I need to report, or is there something I need to
tweak my settings?
Leah Newmark
Capalon VoIP
http://www.olehphone.com
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blame.
I found this thread:
http://forums.digium.com/viewtopic.php?p=49236&highlight=&sid=d3888f3bb90e5c96b5c0432bd632a2d4
but it doesn't help much.
All incoming calls are using IAX.
Did anyone have a similar problem and resolve it?
Thank you.
Leah Newmark
Capalon VoIP
[EMAIL PROT
ain callers (which made me turn on the
record silence option), but my users tell me it's not only those
callers, and sometimes those callers do successfully leave messages; I
only hear when it doesn't work.
What can I do?! I'm stumped, and the situation is intolerable.
Thanks!
dialplan.
I could give you coding examples if you'd like, but I'm not familiar
with Trixbox, and if there is something different about it than a
regular Asterisk system.
Email me if you'd like more help.
Leah Newmark
Capalon VoIP
[EMAIL PROTECTED] wrote:
>
>Message: 1
>D
Yes, she is registered, and her status reads as "ok". She receives calls
fine.
[EMAIL PROTECTED] wrote:
>Message: 13
>Date: Wed, 21 Jun 2006 17:48:34 -0400
>From: "Tom Vile" <[EMAIL PROTECTED]>
>Subject: Re: [Asterisk-Users] Re: User Loses Ability to Make Outgoing
> Calls
>To: "Asterisk Use
We've tried 2 different ATAs, the same hardware, same setup as everyone
else and both have this problem.
This problem is for any number she dials, whether she dials her voicemal
(in-system) or makes a long distance call. She just gets a fast busy,
and I see no output on the console whatsoever.
[E
That's not the problem. The contexts are all fine, and the problem fixes
itself when it feels like it. I am almost positive it's her connection.
The asterisk coding on my end is fine. She has the same setup as the
other 20 employees and they all work fine.
She was running tests on dsltools.com. Is
Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!
Anyway, she *is* able to receive calls. She gets a fast busy when trying
to dial anything.
I know we had her do speed tests on her DSL the end of last year but I
don't remember the outcom
inpoint the problem?
Any assistance is much appreciated.
Leah Newmark
Capalon VoIP
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inpoint the problem?
Any assistance is much appreciated.
Leah Newmark
Capalon VoIP
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whatever the
soundfile is called
exten => s,3,Voicemail(s50) ;s will skip the greeting and just go to the
beep
exten => s,4,Playback(vm-goodbye)
exten => s,5,Hangup
You can then put
exten => 1, Dial(sip/me)
exten => 2, Dial(sip/her)
or whatever your dial statements look like.
Leah Ne
Does adding the line nat=yes into your sip.conf file help?
Leah Newmark
Capalon
www.capalon.com
>Message: 21
>Date: Wed, 28 Dec 2005 11:48:25 +0100
>From: "Rafael Ledesma" <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] Wrong Password?
>To:
>Message-ID:
&g
ys routers. All seem to have the same
problem.)
Leah Newmark
Capalon VoIP
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rankly I'm stumped to where they could
have gone.
Any help would me much appreciated!
Thanks,
Leah Newmark
Capalon VoIP
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I am attempting to set my outgoing calls through H323, however once the call
is picked up, no audio works on either side.
Any suggestions?
H323.conf
--
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=g729
allow=gsm
dtmfmode=rfc2833
gatekeeper = 63.208.156.185
Extensions.
Thanks. I knew this as a possibility, but as for the mailbox owner, he will
then have to record his greetings not through VoicemailMain. That's basically
what I was wondering. I guess Asterisk Voicemail doesn't provide for multiple
extensions per box.
I appreciate your help...and for the person
Is there a way to set up a voicemailbox for multiple extensions with different
greetings, based on the extension dialed? For example, caller1 dials 120 --
he gets u120 or b120 depending on the situation, while caller2 dials 130 and
gets u130 or b130. The catch is that I want the messages to go i
On buyonlinenow.com there's a page sparco.asp...but there is also a little
advertisement on the left for live supportsee if that will get you
anywhere...
> Message: 1
> Date: Wed, 27 Oct 2004 10:14:29 -0500
> From: "Christopher L. Wade" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] [OT] Spa
I am attempting to use the Directory() application. In extensions.conf I have:
exten => 1,1,Directory(default)
and in voicemail.conf I have:
[default]
120 => 1234,Y*** Men***
100 => 1234,D Pal*
118 => 1234,L*** New
121 => 1234,SIP Phone
It works very funny. I press 1, and I get the
I am attempting to use the Directory() application. In extensions.conf I have:
exten => 1,1,Directory(default)
and in voicemail.conf I have:
[default]
120 => 1234,Y*** Men***
100 => 1234,D Pal*
118 => 1234,L*** New
121 => 1234,SIP Phone
It works very funny. I press 1, and I get the
Oh, I think I see...you want to press # so that it will hangup and execute a
transfer to the Asterisk voicemail?
On Thursday 21 October 2004 02:30, you wrote:
> What about using the S(x) command on Dial...after x seconds, when your
> cellphone's voicemail kicks in, hang up.
> I'm not sure I under
What about using the S(x) command on Dial...after x seconds, when your
cellphone's voicemail kicks in, hang up.
I'm not sure I understand why it makes a difference to you whether you hit #
or *...using H is pretty simple, IMO.
>[Asterisk-Users] answer on # key?
From: "M
>
> Message: 10
> Date: Thu, 21 Oct 2004 09:02:16 -0500
> From: "Matthew Boehm" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Press the * key to repeat
> To: <[EMAIL PROTECTED]>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Probably something simple an
>
> Message: 7
> Date: Tue, 19 Oct 2004 15:37:01 -0500
> From: "Matthew Boehm" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Wonderful Success with PAP2-NA
> To: <[EMAIL PROTECTED]>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Finally got "authorized
>Message: 9
>Date: Tue, 19 Oct 2004 14:03:03 -0500
From: "Your Own ISP .com" <[EMAIL PROTECTED]>
>Subject: RE: [Asterisk-Users] How to ring internal extension?
>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
><[EMAIL PROTECTED]>
>Message-ID: <[EMAIL PROTECTED]>
>Content-Typ
Again, this is all speculation, but I've never seen two definitions for a
user...maybe it doesn't know which to use, so it goes to general where the
context is incoming1. Try changing the username for one of the
sip.broadvoices...
>
> Message: 6
> Date: Tue, 19 Oct 2004 14:52:59 -0400
> From: "E
>Message: 16
>Date: Tue, 19 Oct 2004 13:13:52 -0500
From: "Your Own ISP .com" <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] How to ring internal extension?
>To: <[EMAIL PROTECTED]>
>Message-ID: <[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="US-ASCII"
>
>What happens now is this, if
>
> Message: 5
> Date: Wed, 20 Oct 2004 01:10:59 +0800
> From: "Roy" <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] Got SIP response 403 "Forbidden (From header
> is not a Trust host or gateway)" back
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <[EMAIL PROT
> Message: 8
> Date: Mon, 18 Oct 2004 14:32:07 -0400
> From: "Deon Rodden" <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] Voicepulse down for anyone else?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <[EMAIL PROTECTED]>
> Message-ID: <[EMAIL PROTECTED]>
> Content-
I can tell you that you are not alone. It's an issue I believe with Firefly,
and not in your configurations.
>
> Message: 8
> Date: Fri, 15 Oct 2004 13:06:17 +0200
> From: Willem de Groot <[EMAIL PROTECTED]>
> Subject: [Asterisk-Users] FireFly w/ SIP
> To: Asterisk Users Mailing List - Non-Commer
I know that there is a problem when trying to use SIP with Firefly...are you
able to change the option on Firefly to use IAX. It worked much better when
we were trying it out.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Thursday, October 14, 2004 3:10 PM
To:
When I was researching this a bit, I thought I came across a tidbit that said
that this phone is not capable of using "flash"...but now I can't seem to
find that now, and on the contrary, the things I read say that it has such
capability...Right now I am using a regular phone just with a SIP ada
l
as fast forward / reverse (jump forward / jump back). It doesn't seem like
mpg123 can do this. Is there any application that can, that is also compatible
with Asterisk?
Thank you for all your help!
Leah Newmark
Capalon Hosting Solutions
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