Asterisk 1.6.2.20 on Debian Lenny
I'm finding that if no one answers an attended transfer (timeout set by
atxfernoanswertimeout), then the transferrer is handed back to the original
caller, and a beep is played.
In 1.4 I was able to indicate the timeout and failure by setting xferfailsound
to a
Dear Binni;
My asterisk version is:
Connected to Asterisk 1.4.39.1-vici RPM by dem...@goautodial.com
So it is only by 1.4.19?
By the way, the version I am using has been installed using goautodial.
Regards
Bilal
Hi, I've played around with using a database
--- On Thu, 1/26/12, Kevin P. Fleming kpflem...@digium.com wrote:
From: Kevin P. Fleming kpflem...@digium.com
Subject: Re: [asterisk-users] User hit f to disconnect call.
To: asterisk-users@lists.digium.com
Date: Thursday, January 26, 2012, 10:58 AM
On 01/26/2012 07:22 AM, Vieri wrote:
Thanks for help- suggestion fixed the issue
John
On 21 November 2011 11:25, John Taylor j...@vetsurgeon.org.uk wrote:
Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will
get permission to try new firmware later!
JT
On 21 November 2011 10:45, Arthur Stanfield
On 01/28/2012 10:22 AM, Din Assegaf wrote:
Hi All,
I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
But when making A Call from SIP Client, I got cli Warning ... and no
call has been made.
My Sip Client is using lib java peers client http://peers.sourceforge.net/
with
On 01/29/2012 02:34 PM, Mike Diehl wrote:
On Sunday 29 January 2012 8:27:30 am Olivier wrote:
2012/1/29 Mike Diehlmdi...@diehlnet.com
Hi all,
I'm working with the Digium fax for Asterisk product, which is working
pretty
reliably for me.
However, the sendfax application isn't sending status
Hello,
ChanSpy is not completely working for me.
Dialplan :
/exten = _*XXX***,n,ChanSpy(${SIPACC}) ; var $SIPACC has SIP peer
account name/
Verbose :
/[Jan 30 16:25:47] -- Executing [*204***@from-ITEL:10]
ChanSpy(SIP/itel0-2f21, itel1) in new stack
[Jan 30 16:25:48] --
Hi all,
Firstly, apologies if the answer to this question should be obvious.
I have just started working with SRTP and had a self-signed
certificate working perfectly. I have now purchased a CA signed
certificate but can't get it to work properly with Asterisk. I think I
have a configuration
Try
exten = _*XXX***,n,ChanSpy(SIP/${SIPACC}) ; var $SIPACC has SIP peer
account name
Ish
On Mon, 2012-01-30 at 17:04 +0100, Jonas Kellens wrote:
Hello,
ChanSpy is not completely working for me.
Dialplan :
exten = _*XXX***,n,ChanSpy(${SIPACC}) ; var $SIPACC has SIP peer
account name
We mirror off http://packages.asterisk.org to a staging server, then update
from there.
When will this show up on packages.asterisk.org?
Thanks!
EKG
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On 01/30/2012 11:06 AM, Eric Germann wrote:
We mirror off http://packages.asterisk.org to a staging server, then update
from there.
When will this show up on packages.asterisk.org?
Thanks!
EKG
The RPMs should be there in a few minutes.
--
Anyone using the G729 codec to create a h.323 trunk between an Avaya
Communication manager and Asterisk Freepbx System and works? I don't have
the G729 codec installed on the Asterisk and running G711MU on avaya and
getting invalid codec when calling from Avaya to Asterisk.
--
Thanks!
EKG
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker
Sent: Monday, January 30, 2012 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
I have an issue where one of the carriers that my up-line is using is not
offering RFC-2833. I am getting the response from them that if RFC-2833 or
SIP INFO is not offered then I should fall back to inband.
I only have RFC-2833 offered enabled on all phone sets and trunks. The Peer
accounts
I've raised a bug report about this here:
https://issues.asterisk.org/jira/browse/ASTERISK-19268
I'm just wondering who else has been investigating RFC 5922 style
certificate practices?
Which CAs have been able to provide appropriate certificates?
What kind of interoperability testing has
I've just come across this issue:
https://issues.asterisk.org/jira/browse/ASTERISK-17727
I am strongly in support of TLS and I believe this issue will be a
stumbling block for more and more users - because more and more CAs are
using the intermediate certificate chains
For example, the free
On 30/01/12 17:12, Stuart Elvish wrote:
Hi all,
Firstly, apologies if the answer to this question should be obvious.
I have just started working with SRTP and had a self-signed
certificate working perfectly. I have now purchased a CA signed
certificate but can't get it to work properly
I went through the source code and now understand better how dtmfmode=auto
works. In testing I was able to resolve this by setting dtmfmode=auto.
After further testing I will deploy it to production and see if it breaks
anything but I am hoping this will be resolved for the long term.
Thanks
On Mon, Jan 30, 2012 at 7:31 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/28/2012 10:22 AM, Din Assegaf wrote:
The error message is misleading; you are having this problem because the
'm' line in the SDP with the 'audio' offer has a port number of 0 (zero).,
which means it is not
Tonight I tried 4 versions of Asterisk; 10.0.0, 10.0.1, 10.1.0 and trunk.
On 10.1.0 and trunk, I can't successfully enter the password for any
mailbox in voicemailmain on my Aastra 480i phones. All four version
work with a Snom cordless SIP phone. In 10.0.0 and 10.0.1 the Aastra
works
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