Snom make one, it's called the PA1
http://www.snom.com/en/products/sip-paging/snom-pa1/
http://www.provu.co.uk/snom-pa1.html
On Mon, 2010-11-15 at 19:35 +0100, Cassius Smith wrote:
Hi all,
I have had (what I consider) an odd request. The installation I'm
working on now is an office on a
Hi,
our Asterisk is connected to an E1 port. So we are using the
DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for
overlap digits for in-calls? I found the option overlapdial=yes but I
did not try yet. Is that my option? Is there any option for setting an
timeout?
Thorsten
Hi,
Could you please help in getting some info ,
How to configure ring back tones ( CRBT ) with asterisk .
Regards
Mahesh
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Isn't all of that set in indications.conf?
This has been discussed in the past, and I think it is set on a machine
basis, rather than on an extension or call basis.
read, read read! Search and read some more.
Also relate in MUCH more detail what you are tying to do!
John Novack
Govind,
Hello,
I don't really understand how channel Local works. I need that asterisk
initiate a call and get some data (DTMF).
So to do that I've created this dialplan :
; extensions.conf - the Asterisk dial plan
;
[general]
static=yes
writeprotect=no
clearglobalvars=no
[dtmf]
exten
Hi Phuong Hoang,
Without openSER you can use with Asterisk itself for 7 to 10 employees.
Actually what is your workaround, do you want to route the incoming call to
play message then to a queue and finally the call will answered by a team...
like this...?
If yes then you can do very easily by
As I know there are sevearl ways to access MySQL database. The easiest way
is to use Agi server but you can also use Asterisk addons or some other
approach.
Because I'm Visual Dialplan user I prefer Agi option and I use Integration
Server (Agi server) for database access.
It is easy, one block
Local channels behave like an endpoint. So instead of a sip phone
picking up the call, asterisk is picking up the call.
Instead of someone speaking into a sip phone, asterisk can play tracks,
or record digits, etc.
You need to make sure that the call does not end before you're done with
your
Hi,
When refering to Call Waiting feature, I'm thinking about the following
capabilities :
1- multiplexing media with a specific tone, telling one party that another
call is waiting,
2- being able to quickly put an ongoing call on hold and activate the call
already on hold.
A. Has this feature
Thanks Mark for your reply.
I understand all you explained and I agree . All I want to do is abstract for
my app how handle channels (DAHDI or SIP). That's is the main reason, I try to
use Local Channels and NoOp Application so the dial plan should do all the
heavy work (channels
On Tue, Nov 16, 2010 at 6:57 AM, John Novack
jnov...@stromberg-carlson.org wrote:
Isn't all of that set in indications.conf?
This has been discussed in the past, and I think it is set on a machine
basis, rather than on an extension or call basis.
read, read read! Search and read some more.
You can definitely use the local channel and dialplan combination to put
all the work inside dialplan. See my second example on using a local
channel on one side, and dialplan on the other.
In your source leg (dialout) you can write up any logic for handling the
call, you can dial out dahdi,
Hi All -
I pulled from a working system a TDM400 with one s110 fxs and three
x100 fxos. I put it into a new box and the fxs no longer works. The
fxos work just fine. I thought it was odd, but I chalked it up to a
random chance failure and ordered another s110. The replacement
doesn't work
Hi Asterisk Users,
I´m in trouble setting a HA cluster for Asterisk service.
While starting heartbeat service (with *clusterMaster 192.168.1.147 zaptel
asterisk* on haresources), zaptel and asterisk does not start as
I´m expecting, this is the debug result:
ResourceManager[3260]:
On Tue, Nov 16, 2010 at 01:17:08PM -0500, Noah Miller wrote:
Hi All -
I pulled from a working system a TDM400 with one s110 fxs and three
x100 fxos. I put it into a new box and the fxs no longer works. The
fxos work just fine. I thought it was odd, but I chalked it up to a
random chance
Hi. FXS cards use FXO signalling, and vice versa. Think of it this way:
FXS cards want to look like a CO when talking to stations, and FXO cards
want to look like a phone when talking to a CO.
Thanks, Barry. I am aware of this. You'll notice in the config line
that I used fxo signalling.
Has anyone here used T1s with RBS with asterisk?
Cary Fitch
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On 11/16/10 12:55 PM, Noah Miller wrote:
Hi. FXS cards use FXO signalling, and vice versa. Think of it this way:
FXS cards want to look like a CO when talking to stations, and FXO cards
want to look like a phone when talking to a CO.
Thanks, Barry. I am aware of this. You'll notice in the
On Tue, Nov 16, 2010 at 01:55:32PM -0500, Noah Miller wrote:
Hi. FXS cards use FXO signalling, and vice versa. Think of it this way:
FXS cards want to look like a CO when talking to stations, and FXO cards
want to look like a phone when talking to a CO.
Thanks, Barry. I am aware of
On Tue, 16 Nov 2010, Cary Fitch wrote:
Has anyone here used T1s with RBS with asterisk?
Cary Fitch
All of my T1s are RBS. No PRI service here. Am using both Digium and
Sangoma cards.
j
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I'm just thinking that the failure that dahdi_scan see may be because
the s110 isn't getting power.
If you see FAILED in dahdi_scan for the FXS port, then most likely
there will be some indication of what actually failed in the kernel log.
Is there anything in dmesg?
Aha! Thanks, Shaun.
Are other posters getting these annoying messages?
Perhaps serverhallen.com needs to be removed ??
Posting this will generate yet another series of messages from their
postmaster
PITA
John Novack
Original Message
Subject: Delivery Delayed: Re: [asterisk-users] Ring back
So you are using RBS for CO to Asterisk, and SIP for Asterisk to subscriber.
How does the Telco send you the called party info and you to them? DTMF?
What signaling system is being used? EM or?
We are trying to get DS0 EELs from the Telco via T1s and are not sure what
their trunk scheme will be.
On Tue, 16 Nov 2010 14:08:45 +
Vilius Adamkavicius vilius.adamkavic...@invade.net wrote:
For some reason we are seeing Avoiding deadlock for channel in our
Asterisk logs, the logs are getting filled up with an amazing speed
around 12000 lines a second, and all of them are Avoiding
On Mon, Nov 15, 2010 at 1:04 PM, Matt Darnell mattdarn...@gmail.com wrote:
Is this command the best way to access a MySQL database -
MYSQL(Connect connid dhhost dbuser dbpass dbname) ?
I thought I heard that using ODBC was a bit more stable.
Anyone have any experience?
Thanks,
Matt
Hi
I am using Asterisk-1.6.1.6. Recently I had converted billsec and
duration fields in my postgres 8.4 database from integer to numeric. I
wanted to have better accuracy in calculating duration of the call.
When the call is answered I can see better precision (for example
298.758421 sec), The
On Tue, 16 Nov 2010, Cary Fitch wrote:
So you are using RBS for CO to Asterisk, and SIP for Asterisk to subscriber.
In one flavor, yes - our residential dialtone service. We also have PBX
customers that have their own RBS T1 to the CO for their own use. SIP
phones on their desktops of
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, November 16, 2010 2:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 with
On Tue, 16 Nov 2010, Cary Fitch wrote:
Subject: Re: [asterisk-users] T1 with Robbed Bit Signaling
How many T1s per server?
We have never used more than 2 in one server. We don't have very high
traffic levels per location. Multiple people on the list have claimed
multi-hundred
Juan.
Linux User #441131
-- Forwarded message --
From: Juan David Diaz juanch...@gmail.com
Date: Tue, Nov 16, 2010 at 1:38 PM
Subject: HA - asterisk service is not starting
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi Asterisk
On 16 November 2010 22:43, Juan David Diaz juanch...@gmail.com wrote:
Juan.
Linux User #441131
Maybe best on the linux-ha lists...
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Dear asterisk users,
A few weeks ago I've been attacked by a DOS on REGISTER that I've
solved with a fail2ban script.
Now, since a few hours, I have my asterisk 1.4.21.2 running at 100% CPU again.
I've checked the log and it shows nothing related to failed register
or whatever. It just tells me
I also forgot to add that my bandwidth is highly used (mostly out
traffic) since I've detected the attack
On Wed, Nov 17, 2010 at 06:46, Patrick asterisk-us...@ict-synergy.be wrote:
Dear asterisk users,
A few weeks ago I've been attacked by a DOS on REGISTER that I've
solved with a fail2ban
Sounds like your box has been compromised. Check the running processes and lock
down remote ssh access to your server.
Thanks,
--Warren Selby, dCAP
On Nov 17, 2010, at 12:53 AM, Patrick asterisk-us...@ict-synergy.be wrote:
I also forgot to add that my bandwidth is highly used (mostly out
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