2009/4/24 Kevin P. Fleming kpflem...@digium.com
Olivier wrote:
When a SIP hardphone is transfering a call while ringing (caller and
callee don't speak to each other) using phone's Transfer key, it seems
BLINDTRANSFER remains empty.
Though I can see a 302 MOVED TEMPORARILY message coming
Hi,
I am using my own number and not hanging up and audio is also coming
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Thu, Apr 23, 2009 at 9:14 PM, Ruddy Gbaguidi plugwo...@micnes.comwrote:
Maybe the customer hangs up during the AMD analysis or
Hi,
Thanks for your reply
I am using my own number and not hanging up. and sip debug is also not
showing much
information regarding the failure.
please suggest our what might be the problem.
Any help is highly appreciated.
Thanks.
On Fri, Apr 24, 2009 at 4:58 AM, Steve Totaro
Hi,
Thanks for your reply
We are using the Asterisk 1.2.4.
and below the dialplan path. we are orginating the call to
my number and connection it to context cdtest and extension 1.
[cdtest]
exten = 1,1,NoOp( cb amd issue testing )
exten =
2009/4/22 Olivier oza-4...@myamail.com
Hi,
I need to monitor call activity from a custom application software.
The goal is to display things like who is on call or not, who has forwarded
his call to his voicemail, etc ...
When using manager's login command with Event parameter set to on,
Hi,
To further improve Asterisk documentation, would approve manager show
events and manager show event foo commands to be added to CLI ?
Today, it is possible to list available manager commands but not to list
available events, AFAIK.
Regards
___
--
Hi,
Feature originate can be used make call thro' the web. There is a
parameter ,Async, in it. I set it to true but there is no effect.
Actually, I want to do the following.
What I know the function originate is:
originate call --- party A
party A rings
party A answers call
party B rings, party
Hi,
This week (today in fact) Michael Graves talks to Dan Berninger about
the future of wideband VoIP and the upcoming conference. Some of you
might remember the name from a previous conference about FWD. More
about Dan:
Daniel Berninger - Washington, DC based independent technolgy analyst.
hi all,
we do have some troubles with zaptel timing source - we have a setup
with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk
does some handling - calls are leaving on digium card 1 - going to a
siemens hipath - there is some call handling - some of the calls then
are going
2009/4/23 Matt Riddell li...@venturevoip.com:
On 18/04/2009 2:28 a.m., Gavin Henry wrote:
Hi all,
What other open source tools are people using for this? I was looking
at Openfire and their asterisk plugin.
Is it easy to roll your own with res_jabber.so ??
I used openfire in the past, but
Hello,
Well, depending on the version of app_amd that you used when you added
it to Asterisk 1.2, you might need to use HUMAN and MACHINE as the
possible AMDSTATUS instead of AMD_PERSON and AMD_MACHINE. The
AMDSTATUS was changed at some point in the app_amd code, not sure why
they changed it, but
Hello,
I would suggest that you first methodically try every possible
combination of zaptel.conf timing settings(each change follwed by a
hard reboot of the Asterisk server) to see if there is a magic
combination of settings that will work. I don't know if you have the
time for that, or if it
You could use 2 originate commands and connect both of them to a meetme
room?
But surely what you're trying to do is going to confuse the person anyway if
they don't hear anyone when they answer?
Wouldn't it just be better to play a message after party a answers and then
start ringing party b so
Hello,
the issue does occour some seconds after connection the line - but the
hard reboot takes some time...
The cards are TE420 (4th Gen) - version c01a016a.
Zaptel ist 1.4.12.1
The firmware on digium cards can not get flashed - or i am wrong (i have
never heard about that)
regards,
Wolfgang
Hi,
Can someone please help to resolve the followinng issue:
We would like an asterisk user to call a number and when the called party
picks up the phone, we play a message (press 1 to accept call, 2 to reject
call). Only when the called party presses 1, do we bridge the call and the
two parties
Hi all,
I try to install FOP. It's very nice.
In documentation I red that from my dial plan I can launch a popup window
with UserEvent() application.
I try to follow FOP documentation but I can't popup anything. My structure
is:
- server 1: Asterisk system
- server 2: FOP system
- client
On client
Hi,
To further improve Asterisk documentation, would approve manager show
events and manager show event foo commands to be added to CLI ?
Today, it is possible to list available manager commands but not to list
available events, AFAIK.
Regards
The problem is that currently, manager events
Right now, we have a pbx that auto-answers for extension-to-extension calls,
but after the phone has been auto answered, lets the caller press one to
cause the phone to start ringing. (for example, the person's not in their
office so you want it to ring through to voicemail)
I'm able to duplicate
Then a suggestion for the next version would be to have a module which has the
core set of events that are common to most everything for listing and added
too, but still leave it open for the custom events most everyone uses for one
thing or another.
James Shigley
Monroe Telephone Answering
But have you tried to record directly in mp3, without to covert the file?
--- Em qui, 23/4/09, Danny Nicholas da...@debsinc.com escreveu:
De: Danny Nicholas da...@debsinc.com
Assunto: Re: [asterisk-users] Record in mp3
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
you probably don't want to record directly to mp3 as there will be an
overhead in converting the audio on the fly and this will probably break
your call recordings... you should either record in the codec you are using
for phone calls (i think?) or in .wav and then convert afterwards (correct
me
2009/4/24 James A. Shigley j...@answeringserv.com
Then a suggestion for the next version would be to have a module which has
the core set of events that are common to most everything for listing and
added too, but still leave it open for the custom events most everyone uses
for one thing or
Use: console dial
Regards,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Sent: viernes, 24 de abril de 2009 01:07 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 1.6.2
Hi all,
I have not used Asterisk for some time, but decieed to have a go with it again.
I noticed that some commands have been changed, where can one find a list of
them except for the help command?
I want to simulate a phone like I could do in previous versions of Asterisk so
i can type dial
Hi,
exten = 11,1,Playtones(ring)
exten = 11,2,Wait(10)
exten = 11,102,Busy
exten = 11,2,Hangup
this plays me the Ringtone what is set in the indications.conf also over an
iax2 connection to an other Asterisk with SessionProgress(SIP183).
But with this the tone stops:
exten =
The recording I want is to save some reminders that my users can record. It is
not to save a conversation. So I think that there is not an overhead in
converting the audio on the fly in this case.But the question is: Asterisk
suport generate mp3 files directly?
--- Em sex, 24/4/09, Geraint
Im sure someone will correct this if Im wrong Asterisk cant make direct
mp3 records because its not a supported codec. Typically Asterisk records
anything as a gsm, ulaw or alaw file, depending on the codec used to run the
connection to the phone.
_
From:
Usually I used real Digium cards in asterisk systems, so I'm running
into this for the first time.
dahdi_tool reports that dahdi_dummy is in state UNCONFIGURED.
This isn't super surprising, as it seems like the configuration files
for DAHDI are really intended only for configuring real physical
On Friday 24 April 2009 12:40:45 Danny Nicholas wrote:
Im sure someone will correct this if Im wrong Asterisk cant make
direct mp3 records because its not a supported codec. Typically Asterisk
records anything as a gsm, ulaw or alaw file, depending on the codec used
to run the
Jonathan And Dan,
Thank you both for the responses. But the problem turned out to be that - I'm
an idiot. I was placing both calls to our company number for the voicemail
system which I thought was part of a hunt group. As it turns out, only one
call can come in at a time on that number, and
On Fri, Apr 24, 2009 at 01:21:50PM -0500, Tilghman Lesher wrote:
On Friday 24 April 2009 12:40:45 Danny Nicholas wrote:
Im sure someone will correct this if Im wrong Asterisk cant make
direct mp3 records because its not a supported codec. Typically Asterisk
records anything as a gsm,
I'd be really happy if users could use the voicemail menu to change the
number of rings until voicemail picks up.
It seems like the current model of separate Dial and Voicemail commands
would make that difficult, but is there any plan for such a feature in
the future? How about a workaround
Here's a hack to implement this feature
Create global variable RINGS
Use playback and read to let user enter new number
Set RINGS to entered value
Change dial command(s) to include ${RING}
Here's a snippet from my dialplan that changes the default operator
exten =
but you dont really need both parties listen the same message.
You can call the second partie (a new call maybe origiate action) when he
answer you play both the message in each channel if he press one you bridge
the channels.
you first send rings to the caller then you make a new call and play
Adam Moffett schrieb:
I'd be really happy if users could use the voicemail menu to change the
number of rings until voicemail picks up.
It seems like the current model of separate Dial and Voicemail commands
would make that difficult, but is there any plan for such a feature in
the
Deepak wrote:
Hi,
Can someone please help to resolve the followinng issue:
We would like an asterisk user to call a number and when the called
party picks up the phone, we play a message (press 1 to accept call, 2
to reject call). Only when the called party presses 1, do we bridge
the
Anyone knows what should be the configuration of the new solution of
Digium for fax in order to send and receive faxes from PSTN to a fax
machine through an ATA implementing T38 protocol?
___
-- Bandwidth and Colocation Provided by
In order to get 2-way audio without the bridge, you will have to Answer
before Dialing. That's going to mess with your CDR, but you could make this
dial a custom function that you're not going to worry about in the CDR.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Anthony Cascante wrote:
Anyone knows what should be the configuration of the new solution of
Digium for fax in order to send and receive faxes from PSTN to a fax
machine through an ATA implementing T38 protocol?
Fax for Asterisk does not do FAX relay; it terminates or originates FAXes.
--
[top posting continued]
There was an offer, a long time ago, to have all of the prompts for
Zork re-done by Allison in a dramatic reading format, but nobody
(coughSIMONcough) ever got a text file with all the strings
together, and the time to re-write ZoIP to use audio files of each
Has anyone been able to get asterisk 1.6 running under Xen or Amazon EC2?
If yes, can you share your experience please? Is it usable in a production
environment? How is the sound quality? Am I likely to suffer from latency
issues if the extensions are not located in the US?
Any pitfall that I
Have you checked ebay?
Just beware that there are a lot of ATAs on Ebay that are locked to Vonage
or similar providers. While they are not impossible to unlock, it requires
considerable time and good Linux networking experience, as the process
generally involves creating an isolated world (with
Secondarily, MPEG audio compression takes a lot of CPU. Until the last few
years, desktop CPUs weren't even capable of doing realtime MPEG audio
compression, which is necessary if you're going to have the recording ready
by the time the audio input is terminated. Above and beyond that, even
I am running AsteriskNOW as a VM using OpenSUSE 11 and Xen.
Download the .iso to the local disk and point the installation source
at that. Also, make sure to choose full virtualization NOT
paravirtualized for the VM, graphics and the NIC.
I also recommend using a DHCP lease and use a custom
Thanks for the info!!!
2009/4/24 Wilton Helm wh...@compuserve.com
Have you checked ebay?
Just beware that there are a lot of ATAs on Ebay that are locked to Vonage
or similar providers. While they are not impossible to unlock, it requires
considerable time and good Linux networking
When I restart or reboot I can not dial out. The dial() incorrectly
sees dahdi/1 as busy. I call in once from a cell phone, which is
successful then I can call out with out issue. Any ideas would be much
appreciated.
Sangoma B600de
asterisk-1.6.0.9
dahdi-linux-2.1.0.4
Google shows one result for low cost ATA:
http://www.trixbox.org/forums/vendor-forums-non-certified/linksys-cisco/linksys-pap2-and-rt31p2-low-price
Buyer beware! Those are probably counterfeit!
On Fri, Apr 24, 2009 at 19:15, Wilton Helm wh...@compuserve.com wrote:
Have you checked ebay?
Just
Hi,
Thanks for your reply
I have tried the HUMAN as you suggested , but still my problem does not get
solved.
I am answering the call and then running the amd.
Below is the log.
-- AMD: SIP/sip-58ab (null) (null) (Fmt: 4)
Apr 25 00:26:07 NOTICE[27310]: app_amd.c:134 isAnsweringMachine: AMD
c/c'd to the Asterisk list as this is probably relevant to Asterisk as well.
My detailed study of the operation of the 'outgoing' directory reveals that
TXFax() does not delete an expired fax batch file (In the 'outgoing'
directory) until after the end of the dial plan execution.
Is there a
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