OK. I'm getting out the fireproof suit because it's coming and my
hackles have been raised by a number of comments on the list of late.
Disclaimer:
No disrespect intended to the individuals of any *specific* thread. I'm
a little frustrated over energy wasted on pedantic top/bottom posting
Asterisk as a phone system makes perfect sense in a condo. You can get
all the DID's you want and eliminate costs for the owners. You can offer
standard FXO for people who don't care and IP sets for people who want
to upgrade to feature sets.
Your door openner is a piece of cake.
1. Create an
I have 2 FXO channels from which I want to route incoming calls to
different contexts in extensions.conf. I edited the context entries in
dahdi-channels.conf and created matching entries in extensions.conf.
One channel is routed to the new context as I want, but the other
channel is stuck
Hi,
http://store.%50honiceq.com has the quadbri card for $400. We can also
offer free shipping for this card.
The card has 4 BRI ports and is based on the same main chipset
(HFC-4S) as Digium/Beronet/Junghanns cards. It does not have EC
onboard.
best regards
On 1/18/07, Cosmin Prund [EMAIL
I got the same problem with 04/19/05 CVS version. I am using
Grandstream phones. I also noticed that when this happens, an already
hung-up call was still shown as bridged between a SIP phone and a Zap
channel.
On 4/18/05, Shaun Tierney [EMAIL PROTECTED] wrote:
I am currently running
Try to change the macro from
exten = s,1,Dial(${ARG2},20)
to
exten = s,1,Dial(${ARG2},20,${ARG3})
--JJL44
On Wed, 26 Jan 2005 20:35:16 -0500, Steven Frazier [EMAIL PROTECTED] wrote:
I got it to work using the standard example of using exten to ring a phone
vs. the newer macro example.
I
On Tue, 25 Jan 2005 19:33:00 -0500, Steven Frazier [EMAIL PROTECTED] wrote:
Could I ask you a question? You don't have to flash to use the transfer
feature correct or do you? I have tried it both ways and nothing happens.
No, I do not have to flash to use the transfer feature. It works
I had it working. My features.conf file is the same as yours except
for the [featuremap]. I use ## for blindxfer and ** for atxfer.
In my dial plan I use t or T as the Dial() flag.
Make sure that you have beep.gsm and beeperr.gsm in the asterisk sound
file folder. If these files are missing,
a GPL open source
project and asterisk-list is not a Digium support channel. Asking
questions about a vendor other than Digium is no difference than
asking questions about Digium hardware.
--JJL44
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Asterisk-Users
Attended transfer, also called supervised transfer, works like this:
While on conversation with another party, you dial ** the transfer
key sequence. Asterisk says Transfer then gives you a dial tone,
while put the other party on hold music. You dial the transferee
number and talk with the
I justed edited the Wiki Asterisk config file features.conf for this
attended transfer features. Please check Wiki again for details.
Best regards,
--JJL44
On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED]
wrote:
Sorry if I missed the beginning of this thread, but
]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
List
Sent: 20 January 2005 17:28
To: Bruce Komito
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
I justed edited the Wiki Asterisk config file features.conf for this
attended transfer features. Please check Wiki
.
On Fri, 14 Jan 2005 15:08:25 -0800, Asterisk List
[EMAIL PROTECTED] wrote:
I encountered the same problem today. 'lspci -nvv' showed that the
subsystem ID of the TE110p changed from 79fe to 79fa or 797e.
Powering off/on the machine restored the subsystem ID to 79fe and the
wcte11xp module
The current CVS HEAD version already has ## transfer built-in. See
the included configs/features.conf.sample file. You can define your
own transfer key sequence. There is also an attended transfer
feature.
features.conf file:
[featuremap]
blindxfer = ##
atxfer = **
This worked very well for
[inbound]
; This is the list if inbound lines
exten = 2181,1,Answer
exten = 2181,2,Playback(silence/1)
exten = 2181,3,Goto(default,main,1)
exten = 2181,3,Hangup
Notice there are two 2181,3 entries.
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On Wed, 19 Jan 2005 15:55:10 -0700, Jason Kawakami
[EMAIL PROTECTED] wrote:
-Original Message-
My question
concern outgoing calls. How can I configure my extensions.conf to get a
PSTN line on my TDM04B card in the following order : first trying on the
channel 4 then if 4 is busy
I encountered the same problem today. 'lspci -nvv' showed that the
subsystem ID of the TE110p changed from 79fe to 79fa or 797e.
Powering off/on the machine restored the subsystem ID to 79fe and the
wcte11xp module could then load. I already emailed digium support for
this.
On Mon, 20 Dec
David,
I found your post on the Digium archives because I too have noticed
that the flow of traffic on the list has stopped for the past 24 hours
or so. I have replied to many existing threads and started new ones,
only to not see my new messages. I take it from your recent post that
I had a similar problem but not exactly same: when telco lines are
plugged into the FXO ports, initially zap show channel 1 says it is
Onhook but I cannot make outgoing calls. Once I unplug the telco
line and re-plug it, or after there is an incoming call, zap show
channel 1 says Offhook but both
Hello:
I have an asterisk server answering SIP calls.
Whenever a call comes, asterisk answers, plays a gsm file (information) and
dials to another SIP phone.
Using asterisk Master.csv file I only have one record, and don't know if the
second call is answered.
I only know this if:
- The called
Thanks Scott, but I already have a script that creates a call-load.
That's how I have found the OutgoingSpoolFailed error and the other error
with calls being retried without waiting to their retry time.
Has anybody found this problem?
Robert T.
From: Scott Stingel [EMAIL PROTECTED]
Date: Fri,
Hello:
I have found the following problems with outgoing calls with asterisk,
compiled with an updated CVS on 22 Oct.
1.- Problem with retries:
Whenever I set the MaxRetries parameter, to something greater than 0 in a
call-fille, Asterisk ignores the RetryTime parameter and retries every file
Hello:
I need to prepare some detailed stats from asterisk, and I'm asked to show
data I don't know how to obtain it: It's the 'final' number (don't know
what's its name)
In the stats I have to show the caller_id (I have it), the called_id (I have
it) and the final number that actually
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