On Fri, 27 Jan 2017, Michele Pinassi wrote:
i'm using Asterisk as a media box for a VoIP network based on OpenSIPS.
When an user phone is busy, call was forwarded to an asterisk ext:
; ===
; Voicemail on NOT AVAILABLE
; ===
Hi all,
i'm using Asterisk as a media box for a VoIP network based on OpenSIPS.
When an user phone is busy, call was forwarded to an asterisk ext:
; ===
; Voicemail on NOT AVAILABLE
; ===
exten => _VMR_.,1,Noop("from-
Thanks a lot !
I will try the suggested solutions :)
Cheers!
pepesz
On Thu, Jul 26, 2012 at 3:02 PM, pepesz wrote:
> Dear all,
>
> I know the topic comes back like boomerang, but I did not find a nice
> solution.
> Does someone has/knows how to achieve "call back on busy" otherwise called
On 27/07/2012, at 3:42 AM, Richard Mudgett wrote:
>> I know the topic comes back like boomerang , but I did not find a
>> nice solution.
>> Does someone has/knows how to achieve "call back on busy" otherwise
>> called camping?
>> If one is calling the extension and it is busy, then caller should
> I know the topic comes back like boomerang , but I did not find a
> nice solution.
> Does someone has/knows how to achieve "call back on busy" otherwise
> called camping?
> If one is calling the extension and it is busy, then caller should
> get something like "Press 5 to request call back" and a
: asterisk-users@lists.digium.com
Subject: [asterisk-users] callback - disa
Hi/
I am newbe in asterisk.
I try to setup callback with Disa on my home server Anybody help me, pls
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call using the
writecallback.sh above.
Here is a link to a "wait for available" solution -
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pepesz
Sent: Thursday, July
Dear all,
I know the topic comes back like boomerang, but I did not find a nice
solution.
Does someone has/knows how to achieve "call back on busy" otherwise called
camping?
If one is calling the extension and it is busy, then caller should get
something like "Press 5 to request call back" and
Hi/
I am newbe in asterisk.
I try to setup callback with Disa on my home server
Anybody help me, pls
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New to Asterisk? Join us for a live introductory webin
Hello,
I am trying to use a Callback system that return the call to some
number then give it a dial tone with DISA. The callback works well and i
can hear the dial tone, the problem is that DISA doesn't do anything
when i press any extension number of the current context and hangs the
call up
On Mon, 14 Feb 2011 14:21:50 +0100, Gilles
wrote:
>Could it be that while we're in the dialplan after getting a call from
>the FXO, the FXO is just not available until after we exit the
>dialplan?
Made some progress: Asterisk can dial my cellphone if the callback
goes through an SIP trunk instead
On Mon, 14 Feb 2011 10:35:52 +0100, Gilles
wrote:
>If someone's already built a callback like the above using an FXO
>module, I would appreciate any feedback to try and solve this issue.
I learned more about Local channels, but this doesn't work either:
===
[from_fxo]
;Wait for R
On Sat, 05 Feb 2011 12:07:28 +0100, Gilles
wrote:
>I've seen articles about Call files. Is this the easiest way to solve
>this problem?
I'm reading the 3rd edition of the "Asterisk: The Definitive Guide",
but since it's pretty big and there's no guarantee that the answer to
this issue is even in
On Wed, 9 Feb 2011 12:33:00 -0600, Tilghman Lesher
wrote:
>Inotify for spoolfiles is supported starting in Asterisk 1.8.
Thanks for the tip, but I'm stuck with a 1.4 because it must be
patched to run on uClinux :-/
A possible explanation for this issue could be that Asterisk uses
fork() to handl
On Wednesday 09 February 2011 06:28:43 Gilles wrote:
> On Wed, 09 Feb 2011 11:47:09 +0100, Gilles wrote:
> >Unfortunately, I checked how the uClinux kernel was configured for
> >compiling, and the inotify is indeed selected by default :-/
>
> Greping the Asterisk source code for "inotify" only ret
On Wed, 09 Feb 2011 11:47:09 +0100, Gilles
wrote:
>Unfortunately, I checked how the uClinux kernel was configured for
>compiling, and the inotify is indeed selected by default :-/
Greping the Asterisk source code for "inotify" only returned a couple
of hits, in binaries (./main/logger.o and ./mai
On Wed, 9 Feb 2011 00:01:49 -0600, Sherwood McGowan
wrote:
>Nice! That was some good reading!
Unfortunately, I checked how the uClinux kernel was configured for
compiling, and the inotify is indeed selected by default :-/
Linux Kernel Configuration
File systems
[*] Inotify file
Gilles,
Nice! That was some good reading!
On Tue, Feb 8, 2011 at 6:01 PM, Gilles wrote:
> Interesting...
>
> http://en.wikipedia.org/wiki/Inotify
>
> http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/?action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=33403
>
>
>
Interesting...
http://en.wikipedia.org/wiki/Inotify
http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/?action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=33403
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On Tue, 08 Feb 2011 14:23:12 +0100, Gilles
wrote:
>However, by chance, I happened on a pattern: The callfile is handled
>only if I...
>1. Stop Asterisk through its init.d script
>2. Mv the callfile
>3. Start Asterisk through its init.d script
It also works if I launch Asterisk manually with eg. "
Thanks much everyone for the great help. I did go through the last
suggestions about the callfile (no CRLF issue, permissions are 644 and
file owned by root, starting asterisk through strace, etc.), but none
helped.
However, by chance, I happened on a pattern: The callfile is handled
only if I...
>
>
>
> In my (1.4.X) experience, the file just stays in
> /var/spool/asterisk/outgoing and gets “little tags” added until you get the
> problem resolved or delete the file.
>
>
>
That is absolutely true if the file is not processed. I guess he can again
do a "ls -la" in that folder to check permis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood
McGowan
Sent: Monday, February 07, 2011 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback through
*** ***
> If you are sure that permissions are not the problem and you have archive
> set to yes then you can browse the */var/spoo/asterisk/outgoing_done*folder
> to see if the call file is transferred there or not. The file should
> contain some info to help you and it's existence also means
Asterisk runs as root but what about the bash script or the php file that
creates the file? Maybe comment the "mv" command and check the file
permissions by *"ls -la call-filename.call"* to be sure.
*chown root.root call-filename* (if root is really the user running
Asterisk) and then the "mv" com
On 02/07/2011 11:46 AM, Gilles wrote:
Asterisk runs as root, and owns this file as well.
Have you tried setting the permissions of this file to world readable,
to ensure that any user can read it and eliminate potential permissions
problems?
Worth a shot. While you're at it, output from
Real quick, please respond to my question about where the callfile ends up
after a few minutes, as well as the modification time and the permissions on
the file ;-) These are good bits to know
On Mon, Feb 7, 2011 at 10:46 AM, Gilles wrote:
> On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards
On Mon, Feb 7, 2011 at 10:46 AM, Gilles wrote:
> =
> #callfileSIP.call
> Channel: SIP/xlite
> Context: callback-dialtone-auth
> Extension: s
> Priority: 1
> MaxRetries: 2
> RetryTime: 60
> WaitTime: 30
> =
>
Just a thought...
Did you originally generate this callfile on
On Mon, 7 Feb 2011 07:57:07 -0800 (PST), Steve Edwards
wrote:
>> sudo /usr/sbin/asterisk -d -d -d -n -v -v -v
>
>Oops. A '-c' should be in there :)
Thanks Steve for the help.
I launched * with "asterisk -d -d -d -n -v -v -v -c", and ran "module
show" to check that pbx_spool.so is loaded:
==
On Mon, 7 Feb 2011, Steve Edwards wrote:
sudo /usr/sbin/asterisk -d -d -d -n -v -v -v
Oops. A '-c' should be in there :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-386
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan
wrote:
ok, first of all, it can take a little while for those spooled callfiles to
be executed in Asterisk...
On Mon, 7 Feb 2011, Gilles wrote:
Thanks for your help. The same callfile works fine in Ubuntu, but not
at that appliance. Since
On Mon, 7 Feb 2011 04:06:52 -0600, Sherwood McGowan
wrote:
>ok, first of all, it can take a little while for those spooled callfiles to
>be executed in Asterisk...
Thanks for your help. The same callfile works fine in Ubuntu, but not
at that appliance. Since I can dial through the FXO, it doesn'
>
> ... but Asterisk does nothing, altough "show modules" says that
> pbx_spool.so is loaded. Weird :-/
>
> FWIW, Asterisk runs as root, and root owns callfile.call.
>
> Maybe it's the uClinux or the Asterisk I'm using that's configured in
> such a way that callfiles don't work as planned.
>
> Appa
On Mon, 7 Feb 2011 02:59:09 -0600, Sherwood McGowan
wrote:
>That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
>DAHDI/Zap setup... Barring something in your configuration that I don't know
>about, there's no reason that the system should just hang up the call during
>the Wait()
On Mon, Feb 7, 2011 at 2:22 AM, Gilles wrote:
> Lowering it to 5 seconds makes no difference. I also tried adding a
> Hangup before Wait but then the script ends before Wait.
>
>
That's just CRAZY mate! I'm thinking it has EVERYTHING to do with your
DAHDI/Zap setup... Barring something in your co
On Sun, 6 Feb 2011 16:27:33 -0600, Sherwood McGowan
wrote:
>Have you tried playing with the length of the wait? Even if you technically
>need 10 seconds, you could try a lower amount to see if the other priorities
>in that context execute...
Lowering it to 5 seconds makes no difference. I also tr
>From what I can see from your log and the previously supplied snippet of
your dialplan, yes it looks like it's hanging up for a reason other than a
dialplan issue. It definitely doesn't appear to be an issue in the callfile,
since it never gets to the commands that interact with it
Have you t
On Sun, 6 Feb 2011 10:10:06 -0600, Sherwood McGowan
wrote:
>Can you give me/us the output of the full log (verbose set to 5 please)?
>Once I hve that, I can probably help you quite quickly, I work with callfile
>generation often without problem
Here's the output with the console launched with "as
On Sun, 06 Feb 2011 12:05:25 -0500, John Novack
wrote:
>Later Dahdi code may do what you want IF, and only if, your provider
>signals when the call is answered.
Thanks for the information. Telling Asterisk to wait long enough after
I dialed in should be enough.
--
_
Gilles wrote:
On Sat, 05 Feb 2011 16:27:35 -0500, Paul Belanger
wrote:
Easy enough. I would suggest using Disa() for added security.
Thanks for the tip, but then, I would be charged for the call from my
cellphone to Asterisk. I guess it's not a big risk to assume a call
with my C
Can you give me/us the output of the full log (verbose set to 5 please)?
Once I hve that, I can probably help you quite quickly, I work with callfile
generation often without problem
First, you're trying to copy
On Sun, Feb 6, 2011 at 9:36 AM, Gilles wrote:
> On Sat, 05 Feb 2011 12:07:28 +0100,
On Sat, 05 Feb 2011 12:07:28 +0100, Gilles
wrote:
>I've seen articles about Call files. Is this the easiest way to solve
>this problem?
For some reason, Asterisk executes Wait(10), but then hangs up without
running the rest of the commands (cp, echo, mv):
==
[from_fxo]
exten => s,1,Wait(
On Sat, 05 Feb 2011 16:27:35 -0500, Paul Belanger
wrote:
>Easy enough. I would suggest using Disa() for added security.
Thanks for the tip, but then, I would be charged for the call from my
cellphone to Asterisk. I guess it's not a big risk to assume a call
with my CID number is indeed from my c
On Sun, 6 Feb 2011 10:06:42 +0330, Pezhman Lali wrote:
>a2billing also provided call_back daemon, try it
Thanks for the tip, but A2billing requires a LAMP server, which won't
fit on an appliance.
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-- Bandwidth and Colocatio
On Sat, Feb 5, 2011 at 3:27 PM, Paul Belanger wrote:
> On 11-02-05 06:07 AM, Gilles wrote:
> > 2. Asterisk waits until I hang up, calls me back, and prompts me for
> > the number I wish to call
> >
> use exten => h to start a local channel, wait x seconds, dial your cell
> phone.
>
Or you could
Dear
a2billing also provided call_back daemon, try it
best
On Sun, Feb 6, 2011 at 12:57 AM, Paul Belanger wrote:
> On 11-02-05 06:07 AM, Gilles wrote:
> > I'd like to configure Asterisk so that...
> > 1. I ring it from my cellphone with CID number displayed, just to
> > notify Asterisk that I wis
On 11-02-05 06:07 AM, Gilles wrote:
> I'd like to configure Asterisk so that...
> 1. I ring it from my cellphone with CID number displayed, just to
> notify Asterisk that I wish to make a call
>
Easy enough. I would suggest using Disa() for added security.
> 2. Asterisk waits until I hang up, cal
Hello
I'd like to configure Asterisk so that...
1. I ring it from my cellphone with CID number displayed, just to
notify Asterisk that I wish to make a call
2. Asterisk waits until I hang up, calls me back, and prompts me for
the number I wish to call
3. Asterisk puts me on hold through Flash(), w
Look into Call Completion Supplementary Services for Asterisk 1.8
https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29
On Thu, Jan 27, 2011 at 6:48 AM, Harel Cohen wrote:
> Hi All,
>
> I would like to implement a call-back option when called user is busy.
>
Hi All,
I would like to implement a call-back option when called user is busy.
Consider this scenario:
1. A caller is calling a number which is busy on another call.
2. The system will prompt the caller ("press 3 to be called back" etc.) to be
called back when called user is available.
3. Caller h
On Sun, Jan 02, 2011 at 12:04:07AM +, JP CR wrote:
>What I want is when a potential client submits his number... the PBX dials the
>number makes an announcement and dials an extension (which is actually a
>cellhopne dahdi member) and makes the connection.
You might try something based on th
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R
Sent: Sunday, January 02, 2011 3:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback form to place on
On Sun, Jan 2, 2011 at 1:04 AM, JP CR wrote:
> I want to place a form on my site so customers can recieve an mmediate
> callback and the PBX should connect them to a cell sales agent.
>
> Are there anfree modules available for this, or one should code this from
> scratch?
>
> What I want is when a
Greetings,
I want to place a form on my site so customers can recieve an mmediate callback
and the PBX should connect them to a cell sales agent.
Are there anfree modules available for this, or one should code this from
scratch?
What I want is when a potential client submits his number... th
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
Sent: Thursday, August 12, 2010 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback script anyone
Danny Nicholas wrote
Danny Nicholas wrote:
>> From: asterisk-users-boun...@lists.digium.com
>>
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
>
>> Subject: [asterisk-users] Callback script anyone
>>
>
>
>
>> Without diving int
>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J. Oquendo
>Subject: [asterisk-users] Callback script anyone
>Without diving into too many details, does anyone have a simple callback
script that does the following:
>Cal
Without diving into too many details, does anyone have a simple callback
script that does the following:
Caller --> Dial
Asterisk --> "In order to place this call please enter a callback number
to place this call for your pin..."
Caller --> Enters DID to call back for pin
Asterisk --> stores a nu
Hello,
Can anyone give me a sample configuration of Callback feature on a2billing.
Thanks.
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To UNSUBSCRIBE or update options visit:
http://lists.dig
>
>>
>
> Yes, the more expensive ones do. The majority do not.
> Linksys phones.
I have a Snom 320 and an Aastra 480i on my desk, and one of the reasons
I love them (especially the Aastra) is the BLF features.
>
> Its not so much knowing if the user is busy or not, its the ability to
> be auto
> Quick solution that comes into mind:
>
> Set(exten_copy = ${EXTEN});
> Dial(SIP/${EXTEN})
> if ("${DIALSTATUS}"="BUSY") {
> // prompt for camp
> Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
> }
>
> h => {
> Set(call_to=${DB(camp/${exten_copy}/call_to)});
> if ("${call_to}"!="") {
>
t: Re: [asterisk-users] Callback / Camp / Extention Free notify?
Don Kelly wrote:
You're right--he's looking for a camp-on feature, but the campon link that
you found is more of a queuing feature.
A properly-implemented camp-on feature has some advantages.
The caller has full use
Don Kelly wrote:
You're right--he's looking for a camp-on feature, but the campon link that
you found is more of a queuing feature.
A properly-implemented camp-on feature has some advantages.
The caller has full use of their phone while waiting for the call-back. They
can make outgoing cal
pdha...@optusnet.com.au wrote:
Daniel Johnson wrote:
pdha...@optusnet.com.au wrote:
Funnily enough, most people install phones with BLF lamps, on install
something like hudlite/FOP/etc so you know if the person is on the phone
before you call
-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, January 28, 2009 10:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback / Camp / Extention Free notify?
On Thu, 29 Jan 2009, Daniel Johnson wrote:
> pdha...@optusnet.com
> Daniel Johnson wrote:
>
> Jeff LaCoursiere wrote:
> > I think you are looking to use a "campon" feature. Try this:
> >
> > http://www.voip-info.org/wiki/view/Asterisk+tips+campon
> >
> > j
> >
>
> Hi Jeff,
>
> Yes I have seen this feature. Its a half implementation of what we
> requi
> Daniel Johnson wrote:
>
> pdha...@optusnet.com.au wrote:
> > Funnily enough, most people install phones with BLF lamps, on install
> something like hudlite/FOP/etc so you know if the person is on the phone
> before you call them..
> >
> > PaulH
>
> Hi Paul,
>
> Yes I have seen these too
Jeff LaCoursiere wrote:
I think you are looking to use a "campon" feature. Try this:
http://www.voip-info.org/wiki/view/Asterisk+tips+campon
j
Hi Jeff,
Yes I have seen this feature. Its a half implementation of what we
require.
The difference being that you must wait on the phone u
On Thu, 29 Jan 2009, Daniel Johnson wrote:
> pdha...@optusnet.com.au wrote:
>> Funnily enough, most people install phones with BLF lamps, on install
>> something like hudlite/FOP/etc so you know if the person is on the phone
>> before you call them..
>>
>> PaulH
>
> Hi Paul,
>
> Yes I have see
> Daniel Johnson wrote:
>
pdha...@optusnet.com.au wrote:
> Funnily enough, most people install phones with BLF lamps, on install
> something like
hudlite/FOP/etc so you know if the person is on the phone before you call them..
>
> PaulH
Hi Paul,
Yes I have seen these tools. However it is a
pdha...@optusnet.com.au wrote:
Funnily enough, most people install phones with BLF lamps, on install something like hudlite/FOP/etc so you know if the person is on the phone before you call them..
PaulH
Hi Paul,
Yes I have seen these tools. However it is a manual process (simple, I
know)
Funnily enough, most people install phones with BLF lamps, on install something
like hudlite/FOP/etc so you know if the person is on the phone before you call
them..
PaulH
> Daniel Johnson wrote:
>
> Hi,
>
> I am trying to implement the callback feature of our old phone system.
> This fea
Hi,
I am trying to implement the callback feature of our old phone system.
This feature may go by a different name in asterisk?
It worked as follows. If phone A called phone B and it was BUSY, you
press a button to enable a callback.
User A is free to continue work or make other calls.
What t
>
> Hi All,
>
> I'm using A2billing application in order to make callback calls through my
> asterisk server...Everything looks fine except the voice quality...There is
> a lot of cuts in the call with different codecs(G711, and G729)...Please
> note that when making a call from any extensions to
Hi All,
I'm using A2billing application in order to make callback calls through my
asterisk server...Everything looks fine except the voice quality...There is
a lot of cuts in the call with different codecs(G711, and G729)...Please
note that when making a call from any extensions to the same dest
Thank you.
2007/9/12, Atis <[EMAIL PROTECTED]>:
>
> On 9/12/07, Luis Antonio Prata Barbosa <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > Does anybody know if there is a way for a call goes back to transferer
> if
> > unanswered ?
>
> Yes, before Dial to transferer set some variable that have he's
> e
On 9/12/07, Luis Antonio Prata Barbosa <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Does anybody know if there is a way for a call goes back to transferer if
> unanswered ?
Yes, before Dial to transferer set some variable that have he's
extension, and in your defined TRANSFER_CONTEXT, use Dial with g
opti
Hi,
Does anybody know if there is a way for a call goes back to transferer if
unanswered ?
Thanks
Luis A P Barbosa
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Hi,
I have a similar problem:
when I initiate a callback to a local SIP phone no DTMF information is passed
to asterisk.
When the same phone (Grandstream GXP2000) is called from the same asterisk
server with the same settings via an external SIP Gateway (Cisco) DTMF works.
I already tried to use
Hello All,
I don't understand where is the problem...
I have Callback setup and it works fine when tested within US. Works
fine meaning the DTMF tones are passed when prompted to enter the phone
number. But when I test with some international countries, callback
works but DTMF tones are not pas
Greetings,
i've been posted a message to this list in july, which had one response.
Thanks for that idea! Unfortunately asterisk is only a hobby, and did
not have much time dealing with the problem since. My original letter
was long, i wouldn't post it again, the archive url is
http://arch
Are you behind NAT ? Do you have canreinvite=yes ?
- Original Message -
From: "Adam KOSA" <[EMAIL PROTECTED]>
To:
Sent: Monday, June 25, 2007 6:37 PM
Subject: [asterisk-users] callback and bridge problem
> Hi guys,
>
> sorry for the long e-mail, i'
Am Donnerstag, den 28.06.2007, 07:07 +0200 schrieb Adam KOSA:
> Hi guys,
>
> sorry for the long e-mail, i'm only trying to give as much information
> as i think is relevant to my problem (console log, sip.conf and
> extension.conf parts). I've sent this e-mail a couple of days ago, but
> it boun
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts). I've sent this e-mail a couple of days ago, but
it bounced back today.
i've been practicing with callback for a while, but i'm a
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts).
i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.
i have troubles
2007 07:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Callback/ringback
Enclosed bellow is the fragment from extenstions.conf which does two
things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi
Yehavi Bourvine +972-8-9489444 wrote:
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
That's a very nice little script.
--
Warm Regards,
Enclosed bellow is the fragment from extenstions.conf which does two things:
*41 - Does the ring-back staff.
*42 - Calls back the last one who called you.
Regards, __Yehavi:
; regular local extensions:
; The flow is: If not available or no answer send to mailbox if exi
Richard Soderblom wrote:
Hi.
Has anyone had any success in implementing a callback or ringback
function in Asterisk?
I've had a look at the callback-voicemail example on voip-info.org
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
However it won't quite work for me.
I need it for lo
Hi.
Has anyone had any success in implementing a callback or ringback
function in Asterisk?
I've had a look at the callback-voicemail example on voip-info.org
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
However it won't quite work for me.
I need it for local SIP users which most of
Khaled Chehab wrote:
I have an incoming call from pastn number ,the system with deliver it from
e1 .
So I want to close the line an call him .(callback)
This can be done several ways. The receiving operator can pass the call
to a special extension that would either ask for a callback numbe
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback
Khaled Chehab wrote:
>
> Hi dudes
>
> I read a lot of callback tutorials but I failed to make it work, can
> any one tell me how to do it in a brief attached with command line,
> and I w
Khaled Chehab wrote:
Hi dudes
I read a lot of callback tutorials but I failed to make it work, can
any one tell me how to do it in a brief attached with command line,
and I will be thanks full ..
You will need to give us an example of what you want it to do before
that can be done.
Dou
Hi dudes
I read a lot
of callback tutorials but I failed to make it work, can any one tell me how to
do it in a brief attached with command line, and I will be thanks full .
Regards
*
No employee or agent is authorized to conclud
exten => 333,n,Authenticate(1234)
.
.
exten => 333,n+101,NoOp(Is this ok??)
Or i have to explicitly enumerate the priority? ... i'm searching for
doc about this.
as far as i know Auth( ) does not jump to n+101 if you dont use
Auth..(123,j)
enumrations are easier if you use somthing like
Jean-Michel Hiver wrote:
Patricio Valarezo a écrit :
Hi, it's possible to implement a callback without agi?, i'm trying
this but * exits without dialing (if I hungup during s,3 wait) but if
it hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do it
Patricio Valarezo a écrit :
Hi, it's possible to implement a callback without agi?, i'm trying
this but * exits without dialing (if I hungup during s,3 wait) but if
it hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do it? just for learning
Sorr
You can't dial from exten => h
You could use an AGI with a .call file, or you could create the .call
file from inside the Asterisk dialplan. Heck, you could do it with
System() commands. See sample.call in the asterisk source directory. as
well as docs/ in the asterisk source directory.
Pa
Hi, it's possible to implement a callback without agi?, i'm trying this
but * exits without dialing (if I hungup during s,3 wait) but if it
hungs in s,4 it dials, so is there an explanation to this behavior?
there is an alternative to do it? just for learning
thanks for your answers
[followme
- Original Message -
From: "Insider KT" <[EMAIL PROTECTED]>
To:
Sent: Friday, September 01, 2006 1:51 PM
Subject: [asterisk-users] Callback + dtmf problem
Hello all :-)
I am having some problems with my asterisk callback server with mobile
telephones.
When someone from
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