Hmm,
First i must correct myself, MEMBERINTERFACE seems to be NULL, not the “device”
that called in, my bad reading.
Did some changes:
queues.conf
---
[Kinna]
keepstats=yes
ringinuse=no
setinterfacevar=yes
setqueuevar=yes
strategy=rrmemory
timeout=5
wrapuptime=120
extensions.conf
Jonas Kellens jonas.kell...@telenet.be writes:
Hello list,
I'm having some troubles with DTMF tones. When pressing numbers on a Snom
phone, the DTMF-signal takes too long.
Which phone model? If 870, you may want to look at this thread:
http://forum.snom.com/index.php?showtopic=4084
You may
Does it create separet file foreach channel? Or single one?
--
Sent from my iPhone
On Feb 19, 2011, at 12:45 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi Satish,
You can Pass 'r' flag to meetme Application and file will be
recorded nothin to load mixmonitor and other
On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote:
On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed
everything (I think), my Sangoma card initializes right... but there's
no dahdi command -- not from
Satish Patel satish...@hotmail.com wrote:
Does it create separet file foreach channel? Or single one?
--
Sent from my iPhone
On Feb 19, 2011, at 12:45 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi Satish,
You can Pass 'r' flag to meetme Application and file will be
On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote:
On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote:
On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed
everything (I think), my
thanks a lot. that was a problem.
On Fri, Feb 18, 2011 at 8:44 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote:
On Friday 18 February 2011 05:29:56 Borin wrote:
Hello,
trying to load ael module in asterisk ver 1.6.2 got the following:
asterisk*CLI module load pbx_ael.so
Unable to load
On Sun, Feb 20, 2011 at 9:44 AM, Ryan Wagoner rswago...@gmail.com wrote:
On Sun, Feb 20, 2011 at 9:11 AM, Ken D'Ambrosio k...@jots.org wrote:
On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote:
On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote:
Hi, all. I've finally made
On 2/18/11 5:18 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-02-18 03:59 PM, Cassius Smith wrote:
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I
On Sat, Feb 19, 2011 at 04:15:15PM -0500, Ken D'Ambrosio wrote:
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed
everything (I think), my Sangoma card initializes right... but there's no
dahdi command -- not from the base, nor as a subset of the core
commands. I've got my
You -- as usual -- hit the nail on the head; I'd actually figured it out
at probably roughly the same time as you e-mailed, because I bumped into
this:
Asterisk Module and Build Option Selection
[...]
XXX chan_dahdi
[...]
DAHDI Telephony
Depends on: dahdi(E), tonezone(E)
Can
It's been my experience that the MEMBER... Variables are populated by the
person who answers the queue call. If no one answers the call, I would imagine
the variables would be null.
Thanks,
--Warren Selby, dCAP
On Feb 20, 2011, at 2:17 AM, magnu...@inputinterior.se wrote:
Hmm,
First i must
*Bump* No takers? Perhaps no-one else thinks this is a bug?
Regards,
Steve
On 7 February 2011 16:45, Steve Davies davies...@gmail.com wrote:
Hi,
The following IAX config (slightly edited) causes an issue for me in
version 1.6.2.16.1, where my CDR data is unreliable.
[user1]
type=friend
On 2/02/11 7:05 AM, Olivier wrote:
Hi Matt,
Too bad I can't be more helpful on this but could work around this issue ?
Nah, in the end I just learnt how to use LCR with mISDN.
I upgraded DAHDI, LibPRI, Asterisk to latest versions and still no go -
although the errors stopped happening.
On 11/02/11 6:54 PM, William Stillwell wrote:
I was getting unable to make channel..
We couldn't get it to work properly until we upgraded to Asterisk 1.8 at
which stage it magically started working (with the same configs etc).
--
Cheers,
Matt Riddell
I was also informed it only works in 1.8, I think there was a protocol
change I think that wasn't back ported to 1.6.
Also here:
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
Calling using Google Voice or via the Google Talk web client requires the
use of Asterisk 1.8.1.1 or
And confirmed, just upgraded to 1.8.x.x branch, outbound/inbound working
fine.
Now, no outbound callerid? Shows 'unknown caller' on called party's
handsets.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf
Thanks for the insight will try that one.
Is there any suggested value for this parameter that best work well on fax?
Regards,
Mac
From: Moises Silva moises.si...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February
Unknown Caller most likely refers to the CID Name, CID Number should
be provided as your Google Voice number.
On 2/20/2011 5:53 PM, William Stillwell wrote:
And confirmed, just upgraded to 1.8.x.x branch, outbound/inbound working
fine.
Now, no outbound callerid? Shows 'unknown caller' on
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Sunday, February 20, 2011 10:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Gtalk/Jabber Issue
William,
It still looks like something is not properly set with your account on
Google Voice. Have you had a chance to follow the recommendations I
gave you earlier in the thread?
If the account is properly set the dial string will need to look like
this,
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