Looks like version 11.3 did not fix my issue.
http://pastebin.com/gd291Bqz
On Thu, Apr 4, 2013 at 1:23 PM, Duane Larson wrote:
> Thanks Jim. Searched through the change log for "deadlock" but nothing
> really stuck out. I'll upgrade to 11.3 and see if that makes a dif
Thanks Jim. Searched through the change log for "deadlock" but nothing
really stuck out. I'll upgrade to 11.3 and see if that makes a difference.
On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas wrote:
> On 04/03/2013 08:15 PM, Duane Larson wrote:
>
>> So it just happene
version 11.2.1
http://pastebin.com/mbjSSAWM
This has to be a bug right? I am thinking of opening an issue on the
Asterisk JIRA system
On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson wrote:
> It just happened again on the 11.0.1 box and I was able to grab a debug.
> I am hoping someone ca
It just happened again on the 11.0.1 box and I was able to grab a debug. I
am hoping someone can tell me if this is a bug or something wrong with my
config.
gdb asterisk-bin/sbin/asterisk 29048
Go here for the debug output
http://pastebin.com/DGXx0BSk
On Tue, Apr 2, 2013 at 7:42 PM, Duane
I am currently running two different versions of Asterisk
11.0.1
11.2.1
I have noticed the bug occur on both servers.
The issue is that when I try to dial a phone number sometimes the call will
never go out. I will check the Asterisk server with NGREP and see that the
SIP messages are making it
ideas?
On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson wrote:
> One more piece to add. I had mentioned before that I could get a call from
> a PSTN user to work the first time. So here is all the output of a Good
> call from a PSTN user after I have performed a "RELOAD&qu
this call up all calls from the PSTN
afterward get put in the queue automatically and the agent never gets
called.
On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson wrote:
> Ok. Something isn't right. With a user that is local to my SIP user
> database calls the queue phone number eve
; Next time when this happens, try 'core show channels' to check whether
> Agent channel is released or not.
>
> [SATISH]
>
>
> On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson wrote:
>
>> Yesterday I rebooted the server and it seems to be working again. Not
>>
Yesterday I rebooted the server and it seems to be working again. Not sure
what the reboot might have changed. Hopefully it doesn't happen again but I
can't be sure. To answer your question I have the sip.conf in my mysql
database and in MySQL I have callcounter set to yes. I don't have a colum
Queue not sending call to Agent
I am having an issue and i am not sure if it is a bug or a config issue. I
was originally running Asterisk 1.8.1.1 when I noticed this issue. I
upgraded to 1.8.4.2 to see if that would fix it but it didn't.
The issue is that I have a call queue and the agen
Patch worked like a charm. Thanks Chad. Thought I had done something wrong
when installing. Really appreciate it.
On Mon, Jan 10, 2011 at 2:27 PM, Duane Larson wrote:
> Thanks Chad. I will try the patch.
>
>
> On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace > wrote:
>
>
Thanks Chad. I will try the patch.
On Mon, Jan 10, 2011 at 2:27 PM, Chad Wallace
wrote:
> On Sun, 02 Jan 2011 17:44:19 +
> duane.lar...@gmail.com wrote:
>
> > I have asterisk 1.8.0 installed and I am not able to forward a
> > voicemail from one users mailbox to another user.
>
> I had the sa
I still can't figure out why this isn't working. I updated to the latest
version of Asterisk 1.8.1 with no luck. I am using Realtime for sipusers
and vmusers if that makes any difference. I tested this on a new install
and saw the following
under the folder where I installed Asterisk I had
/hom
I have asterisk 1.8.0 installed and I am not able to forward a voicemail
from one users mailbox to another user.
I have the user log into their mailbox
press 8 to forward a message
enter the extension of the user I wish to forward too
I don't prepend a audio message
and press # to send the mess
I have configured my mysql database by following this link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue
The only difference is that I am using ODBC instead of MySQL with Realtime.
Within extensions.conf I have the following for my queue
exten => 9**2**1611,1,Answer
exten => 9**2**1
Snom
Sent from Droid
On Dec 17, 2010 12:36 PM, "Matt" wrote:
I'm looking for a wireless desktop VoIP phone. Does any exist?
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us f
Awesome. Didn't notice that, but that is my fault for not reading the
changelog or the updated sample configs. I will try this out.
Thanks all for the comments.
On Wed, Dec 1, 2010 at 10:09 AM, Tilghman Lesher wrote:
> On Tuesday 30 November 2010 18:34:17 Duane Larson wrote:
&g
Thats sounds interesting too. I will look into that also.
On Wed, Dec 1, 2010 at 1:30 AM, Stefan Schmidt wrote:
> Am 01.12.10 05:10, schrieb Duane Larson:
> > For me OpenSIPS will do most of the work. Asterisk will only handle Hunt
> > Groups/Queues, IVRs, and Voicemail when Op
ou have automated failovers in
> MySQL in the 1-2 second range.
>
> Singer
>
>
> On Tue, Nov 30, 2010 at 19:51, David Backeberg wrote:
>
>> On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson
>> wrote:
>> > I have MySQL Cluster set up for OpenSIPS which allo
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime I am
only able to point to a single IP address for a database. If that databas
Your router would have to do per-destination when it came to load balancing
between the two dsl circuits. That way a single call could only use one dsl
path.
On Nov 2, 2010 7:36 PM, "Dan Journo" wrote:
Hi,
I've got a client with two ADSL connections for redundancy.
Is it possible to set
I am trying to set up Hunt Groups and I am having some issues. Here is what
I am trying to do. All my users actually register with OpenSIPS. Asterisk
is using Realtime and I have set up a MySQL View Table so that Asterisk
see's all the SIP users info that OpenSIPS has. This is what I have
confi
I currently have OpenSIPS set up with users and most of my call handling.
OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD,
etc. So I want to send these types of requests to Asterisk. I also want to
set Asterisk up as Multi Tenant. So my question is
How can I send requests
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