Hi everyone,
when i use the automated dial out,I found that once the zap answerd,the
contex will be exectued, but i don't hope do it ,i hope when extern phone
answered ,then ,the context will be exectued.
Anyone can help me solve the problem!
the call file is:
Channel: Zap/g0/15015895665
Conte
I have one ST2030 bought for testing. Indeed it has a very intuitive user's
interface, bue I've found two drawbacks:
- Its sound quality has some place to be improved...
- It has no RPID support (displaying the name of the called party).
If these two issues are fixed, then it might be the b
I would like to thank everyone for their input. This project is completed and
the solution is working wonderfully! There was some mention of some people
having difficulties with asterisk/mssql connectivity via the dialplan when
under heavy load. Running the connection through agi was the suggest
On Thursday 18 December 2008, Justin Phelps wrote:
> I've been struggling with an ongoing problem the last month.
>
> Here is the layout of the wiring:
> T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank
>
> (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server
> zap c
Jose P. Espinal schrieb:
> Any Idea why these return values are not 'officially' documented somewhere ?
The official documentation is in the code,
enum ast_extension_states in include/asterisk/pbx.h:
---cut---
/*! \brief Extension states
\note States can be combined
- \ref AstExt
On Fri, Dec 19, 2008 at 12:08, David fire wrote:
> set(CALLERID(number)=000)
> David
Keep in mind that with doing that, you would loose the caller ID
number for the CDR -- thus there will be no record of the caller ID
anywhere (asterisk-related, at least).
I believe if you use a Local channel (D
Hi,
The queue in my asterisk is work fine, but when anyone is waiting for
contact if type one key in phone the asterisk answers.
How can I disable that?
Thanks in advance
http://www.datasul.com.br/tec/2008/Techequipes/Infra/Ass_email/img/lateral.j
pg
Geraldo Coelho
Consu
Hi
i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx,
So anybody can recah me by dialing my SIP uri. same time my DNS on same
server where currently Asterisk running.
how ican implement this. Please help me with config details at DNS &
Asterisk point of view. anybody can pro
On Mon, Dec 22, 2008 at 16:30, amit salunkhe wrote:
>
> i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx,
> So anybody can recah me by dialing my SIP uri. same time my DNS on same
> server where currently Asterisk running.
> how ican implement this. Please help me with con
The problem I was experiencing is still occurring, and it is getting
worse. There are several names that Festival "gets stuck on". I don't
know if it is a Festival problem or an Asterisk problem. The scenario, a
call comes in goes through the dialplan (shown below in original message),
and eithe
Thank you very much Phillip,
Any Idea why these return values are not 'officially' documented somewhere ?
Regards,
--
JPE
Philipp Kempgen wrote:
> Jose P. Espinal schrieb:
>
>
>> Until now everything is going Ok but something a little (in my oppinion)
>> strange is going on with the 'Ex
Jose P. Espinal schrieb:
> Until now everything is going Ok but something a little (in my oppinion)
> strange is going on with the 'ExtensionState' command;
> The problem is that it does not returns the 'Status' as it's suposed to,
> mentioned in the A.T.F.O.T book - version 2.,
> where it sais
Hi Gang,
I'm trying to make an application to upload a tiff via a web
interface, slap a cover page onto it, merge the two into a new tiff and send
it out via txfax. I'm able to get it out to a fax machine using this
sequence:
/usr/bin/tiff2pdf /tmp/faxout/1229978819_filea.tif -o
/tm
Not a solution, but a work-around. You could write a routine in Perl or C
or something to monitor the database and send out the voicemail via IMAP
when it comes in. I do a similar thing to increase the volume of a received
voicemail (WAV) and send it out to an Iphone.
-Original Message-
Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be
useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP / realm /
sip user), you just need to setup the HTML link as that: (Attached is the HTML
page example)
/***
Hi All -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem seems to
be with libc-client. Specifically, asterisk is not able to access the
mm_dlog f
Hello List,
I have been working on a PHP application in order to build a BLF style
script.
Until now everything is going Ok but something a little (in my oppinion)
strange is going on with the 'ExtensionState' command;
The problem is that it does not returns the 'Status' as it's suposed to,
me
Hi all,
Sometimes when making a PC to PSTN call through asterisk, I got no audio in
both sides...tracing by wireshark, I can find that RTP packets are hitting
my PC but no audio...Can someone guess what could be that issue?
Maybe it's a latency issue?
Regards
_
One way to do this would be using
func_odbc.conf
This allows you to define dialplan functions that are based on ODBC
queries.
Like this, which looks up a meetme room number based on the project
and the 'space' number within that project (sub-project if you like).
[SPACE]
prefix=MEETME
dsn=my_or
Fred Posner wrote:
> Yeah, they finally updated via their twitter account... Seems a
> generated exploded.
>
> http://www.voiptechchat.com/voip/165/voip-carrier-voicepulse-suffers-outages-uses-twitter/
>
>
> Fred Posner
> www.teamforrest.com
>
>
> On Dec 22, 2008, at 11:15 AM, Shane Young wrote:
We sell BT200's premodified for this exact purpose. We install an RCA/Phono
jack on the back so you can connect the phone directly to your overhead paging
equipment. They also have a toggle switch for moving between the chassis
speakerphone and the external jack. We're currently using these in a
On Mon, 22 Dec 2008, Jerry Geis wrote:
> Is there an ATA type device out there that has low level audio out for
> connecting speakers?
>
> My asterisk server is in one building, I wish to have speakers in
> another building
> and connect them up to a low level audio device that I can call into and
Look at Valcom or Viking. They make the paging hardware hat interfaces with
FXO or FXS
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Jerry Geis
> Sent: Monday, December 22, 2008 11:22 AM
> To: aste
Yeah, they finally updated via their twitter account... Seems a
generated exploded.
http://www.voiptechchat.com/voip/165/voip-carrier-voicepulse-suffers-outages-uses-twitter/
Fred Posner
www.teamforrest.com
On Dec 22, 2008, at 11:15 AM, Shane Young wrote:
> Quoting Fred Posner :
>
>> Starti
Is there an ATA type device out there that has low level audio out for
connecting speakers?
My asterisk server is in one building, I wish to have speakers in
another building
and connect them up to a low level audio device that I can call into and
speak.
Can I connect speakers into the FXS or
Quoting Fred Posner :
> Starting around 10:00 AM EST.
>
> All services from them whether I connect by IP or DNS (both east coast
> and west). Anyone else?
Yes, I'm experiancing the same problem.
Their www.voicepulse.com and connect.voicepulse.com seem to be offline
as well.
--Shane
-
Starting around 10:00 AM EST.
All services from them whether I connect by IP or DNS (both east coast
and west). Anyone else?
Fred Posner
f...@teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
__
forums - sigma wrote:
> having deployed a fair amount of phones I have the following observation
> (and these observations are worth what you paid for them :-) )
>
>
> 1. Linksys 942, my preferred mainstream desk phone, a bit more expensive
> than the Polycom IP330. Be careful as there are two SK
2008/12/22 David fire
> hi
> this is the situation i have a queue and i am monitoring when an important
> customer call i want to take it off from the queue and send it to an agent
> directly can i do that? how?
> thanks
> David
>
>
> --
> (\__/)
> (='.'=)This is Bunny. Copy and paste bunny into
Hey everyone,
A while back I worked on a project to measure call quality. I've
finally gotten around to releasing it and I'm calling it recqual (Real
Call Quality). There isn't much to it and it should be considered
alpha quality. I'm hoping some of the bright minds on the list can
help me o
Hi all,
I know I'm probably stirring up a hornet's nest with this question/comment
but I've spent the last few days working on a PHP-based class for the
manager interface as we're preparing for a pretty big upgrade at our call
center and I'm revamping all of the management apps I've written. I c
This article might prove useful.
http://en.wikipedia.org/wiki/Iptables
telnet could provide the missing piece(s) of this puzzle for you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk
Sent: Frid
What does your extensions.conf look like for this call? If you can insert a
ww into your Dial command (ie, change 18005551212 to ww18005551212) this may
improve your dialing behavior.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On B
Rob Hillis wrote:
> Michael wrote:
>> My experience with Grandstream is that are one of the better 'cheap' ones,
>> but
>> cheap non the less.
>
> I am yet to run into a worse IP phone than the Grandstreams - although
> having said that, I should say that I've always steered clear of most
> of
hi
this is the situation i have a queue and i am monitoring when an important
customer call i want to take it off from the queue and send it to an agent
directly can i do that? how?
thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
(")_(")signature to help him gain wor
On Tue, 23 Dec 2008 00:06:19 you wrote:
> I'm not sure mate - as I don't use HylaFax on that particular server.
> Hopefully, someone else can help. Sorry.
I have installed the pre-requisite versions asked for and successfully got AGX
addons installed. Except it doesn't work. "Failure to train re
Dear All,
When one configures a standalone asterisk voicemail and attach to a legacy
PBX, and the PBX transfers a busy, no-response or switchedoff extension to
Asterisk. What would be the source and destination caller IDs that would get
to asterisk voicemail?
Thanks,
...
Hi,
To force user to behave correctly, i want to make a process of disconnecting
every member but one (special alarm phone) every day at a special time.
I'm thinking of a cron job that will create a "call file" that will dial an
appropriate extension to do the job, is this the correct way ?
I'm
"Richard Wurman" writes:
> ...but currently has been removed. Does this imply the issue is
> resolved? Of course, I can always try it out, I just want to verify if
> there is a definitive answer.
Digium's drivers used to have issues with receiving extra interrupts.
Fortunately Digium has seen th
I'm not sure mate - as I don't use HylaFax on that particular server.
Hopefully, someone else can help. Sorry.
-->> -Original Message-
-->> From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-->> boun...@lists.digium.com] On Behalf Of Michael
-->> S
On Mon, 22 Dec 2008 23:46:28 Andrew Thomas wrote:
> ...described in the README file ;).
>
> SpanDSP - 0.0.4 pre 16
> LibTiff - >= 3.8 but <4.0
>
>
> I had to trawl around for the right SpanDSP - but I can e-mail a copy to
> whomever wants one (drop me a personal e-mail and I'll attach it by return)
...described in the README file ;).
SpanDSP - 0.0.4 pre 16
LibTiff - >= 3.8 but <4.0
I had to trawl around for the right SpanDSP - but I can e-mail a copy to
whomever wants one (drop me a personal e-mail and I'll attach it by return)
HTH
Andy
-Original Message-
From: asterisk-users-bo
Hi,
2008/12/22 Andrew Thomas
> You don't really need to use any local MTA if you use the sendEmail
> script.
>
> I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/
>
> This actually works by 'talking' directly to any SMTP server - even
> remote ones (I use our Exchange serve
Is this package capable of receiving faxes where the up line connectivity is
FoIP through a T38 SIP connection?
Michael
___
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To UNSUBSCRIBE or update opti
Hi Andrew,
2008/12/22 Andrew Thomas
> JFYI - I run (successfully) agx-addons with 1.4.22 and Etch.
>
> Make sure you have the right version of SpanDSP installed (as well as the
> tiff libraries).
which are (thinking of both SpanDSP and libiff) ?
>
>
> -->> -Original Message-
> -->>
JFYI - I run (successfully) agx-addons with 1.4.22 and Etch.
Make sure you have the right version of SpanDSP installed (as well as the tiff
libraries).
-->> -Original Message-
-->> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
-->> boun...@list
You don't really need to use any local MTA if you use the sendEmail
script.
I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/
This actually works by 'talking' directly to any SMTP server - even
remote ones (I use our Exchange server for our e-mails).
HTH
Andy
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