Hi All,
Asterisk Version: 1.8.13.0
CentOs : 6.3
Continues getting this error while submitting cdr record.
[Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic
number 0x1 for 0xb61ddeb8
[Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic
number 0x1 for
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote:
Hans,
I did not try 10 or 11 as I run 1.8.15. Following are the related
conf files.
gtalk.conf
[General]
context = default
allowguest = yes ; Required if you want to accept calls
from people Not on your contact
Also you could have a look at openfire and it's Asterisk-IM plugin
On Wed, Sep 12, 2012 at 10:41 AM, Hans Witvliet aster...@a-domani.nlwrote:
1.8 machine
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Moving from 1.4 to 11 deadagi is deprecated.
Is there a suggest way in the dialplan to handle the case of either.
Was hoping to keep one extensions.conf file and just call the
appropriate agi/deadagi.
What's the best way to do that?
Jerry
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I am looking for any information of the possible use of Asterisk in any of the
systems used by/for the London Olympics.
Please get in contact if you know of any such use(s).
Thank you,
David
Digium logo
David Duffett
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close,
Any suggestions? Or I should address it to asterisk-dev maillist as feature
request.
Regards,
Pavel
Hi David,
For sure I will ask them but I think Asterisk should be able to handle this
case because it doesn't matter if it is adhearsion or something else. If it
is not present there is no way to
Try adding qualify=yes
Eric
I added this and I still had one time last night at 3am that
said Unspecified.
Is there something else?
I put it in the [general] section of sip.conf on both machines.
Jerry
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I have a server with an asterisk ss7 link connected to a Siemens working
well for over a year.
A few days ago I started having problems with signaling.
I found the following logs in / var / log / messages
Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to
TE2/0/2
Sep 12
Sorry, in the last line I try to say ...
With asterisk stopped, and only driver dahdi up, messages still appear...
On Wed, Sep 12, 2012 at 12:19 PM, equis software equissoftw...@gmail.comwrote:
I have a server with an asterisk ss7 link connected to a Siemens working
well for over a year.
A
On Wed, Sep 12, 2012 at 12:19:41PM -0300, equis software wrote:
I have a server with an asterisk ss7 link connected to a Siemens
working well for over a year.
A few days ago I started having problems with signaling. I found
the following logs in / var / log / messages
[1018427.030959]
Thanks, I'll try changing cables.
On Wed, Sep 12, 2012 at 12:21 PM, Shaun Ruffell sruff...@digium.com wrote:
On Wed, Sep 12, 2012 at 12:19:41PM -0300, equis software wrote:
I have a server with an asterisk ss7 link connected to a Siemens
working well for over a year.
A few days ago I
Hi All;
Is there a module (addon or already built in) that enable us to receive the fax
on the telephony card and save it as image (or any other format) and sent it to
email?
Regards
Bilal
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FFA (Free Fax Asterisk) saves the incoming fax as a TIFF which you can
email.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Wednesday, September 12, 2012 5:01 PM
To:
On 10/08/12 18:38, Chad Wallace wrote:
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcuss...@open-t.co.uk wrote:
I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP calls are crystal clear, but all
calls bridged to POTS have a significant
On 9/12/2012 5:33 PM, Sebastian Arcus wrote:
On 10/08/12 18:38, Chad Wallace wrote:
On Tue, 31 Jul 2012 09:44:26 +0100
Sebastian Arcuss...@open-t.co.uk wrote:
I have two setups with SIP hardware phones as extensions and POTS
lines as trunks. Internal SIP to SIP calls are crystal clear, but
On 9/12/2012 1:41 AM, Hans Witvliet wrote:
On Wed, 2012-09-12 at 00:01 -0500, Vladimir Mikhelson wrote:
Hans,
I did not try 10 or 11 as I run 1.8.15. Following are the related
conf files.
gtalk.conf
[General]
context = default
allowguest = yes ; Required if you want to
I know that asterisk on virtual machine require a timing source. What would
you suggest to use for timing? We will plan to use only SIP and IAX2.
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New to
On Wed, Sep 12, 2012 at 11:52:40PM -0400, Mark Robinson wrote:
I know that asterisk on virtual machine require a timing source.
What would you suggest to use for timing? We will plan to use only
SIP and IAX2.
If you're on a newish kernel (something later than v2.6.22), can use
app_confbridge
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