Hello,
I'm using py-Asterisk 0.5.3.
I'm trying to use it along a Tkinter-based GUI so I've dedicated a thread
for reading incoming AMI events.
Which is the preferred way to gracefully exit from an event loop ?
More precisely:
This thread is waiting for input events with Manager._read_packet()
Hello,
I'm using MySQL realtime Call Queues (table /queues/ and table
/queue_members/).
I would like to ring the members of the call queue in a certain order.
Therefore I use ring strategy /lineair /and I put the members into the
table /queue_members/ in the order in which they have to be
Hello,
Has someone successfully configured support of either callto or tel
protocol in MS Office in general or MS Office Online's Outlook specifically
?
(I'm referring here in Outlook client embedded in MS Office cloud service).
If positive, what are the basic steps to enable such feature
On 11/02/14 18:45, Dave Platt wrote:
Just checking the transcoding on our Asterisk boxes and I get the
following results.
I have the g726, ilbc and lpc10 formats and codecs enabled in 'make
menuselect' so I dont understand why its showing as no translation path.
Any ideas?
Are the modules
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, February 12, 2014 3:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime Call Queues : call members in certain
Hello,
Getting this error on dahdi_cfg. Reverting to 2.8 the error goes away:
*line 15: Unable to create 'dahdi_cfg' mutex.*
Is this a problem?
Thanks in advance.
Full detail:
[ebox] dahdi_cfg -vvv
DAHDI Tools Version - 2.9.0.1
DAHDI Version: 2.9.0
Echo Canceller(s): HWEC
Configuration
On Wed, Feb 12, 2014 at 05:21:24PM +, Rodrigo Borges Pereira wrote:
Hello,
Getting this error on dahdi_cfg. Reverting to 2.8 the error goes away:
*line 15: Unable to create 'dahdi_cfg' mutex.*
Is this a problem?
Thanks in advance.
Full detail:
[ebox] dahdi_cfg -vvv
Hello Shaun,
This system is a custom distro, based on debian and currently built around
kernel 2.6.35.8. Is sem_open introduced only in DAHDI 2.9 ?
On Wed, Feb 12, 2014 at 5:37 PM, Shaun Ruffell sruff...@digium.com wrote:
On Wed, Feb 12, 2014 at 05:21:24PM +, Rodrigo Borges Pereira
Hello,
How does extensions.lua compares to extensions.conf or extensions.ael on
stability, performance and features ?
Would you recommand extensions.lua as an easy/easier way to access
memcached, redis or equivalent ?
Thoughs ? Comments ?
Regards
--
Maybe the problem is lack of tmpfs?
statfs(/dev/shm, 0xbff88568) = -1 ENOENT (No such file or
directory)
Is this a new requirement for DAHDI?
On Wed, Feb 12, 2014 at 5:49 PM, Rodrigo Borges Pereira
rodrigoborgespere...@gmail.com wrote:
Hello Shaun,
This system is a custom
On Wed, Feb 12, 2014 at 05:49:49PM +, Rodrigo Borges Pereira wrote:
Hello Shaun,
This system is a custom distro, based on debian and currently built around
kernel 2.6.35.8. Is sem_open introduced only in DAHDI 2.9 ?
Yes. It was added in [1] in order to prevent errors when multiple
Hi all,
I've got a customer who's reporting ghost calls. Essentially, the phone
rings, they pick up, and there's no body there.
It is NOT one-way audio, and it doesn't happen all the time.
We use voipmonitor to watch calls, and this is what we saw for the call in
question:
| calldate
On Wed, Feb 12, 2014 at 05:55:41PM +, Rodrigo Borges Pereira wrote:
Maybe the problem is lack of tmpfs?
statfs(/dev/shm, 0xbff88568) = -1 ENOENT (No such file or
directory)
Is this a new requirement for DAHDI?
tmpfs is not a requirement per-se, but it's how most POSIX
Well, I created /dev/shm and mounted tmpfs on it, and the problem is no
more. Just creating /dev/shm was not enough.
thanks.
On Wed, Feb 12, 2014 at 6:06 PM, Shaun Ruffell sruff...@digium.com wrote:
On Wed, Feb 12, 2014 at 05:55:41PM +, Rodrigo Borges Pereira wrote:
Maybe the problem is
Just one last question: do you have another suggestion about this?
thanks.
On Wed, Feb 12, 2014 at 6:07 PM, Rodrigo Borges Pereira
rodrigoborgespere...@gmail.com wrote:
Well, I created /dev/shm and mounted tmpfs on it, and the problem is no
more. Just creating /dev/shm was not enough.
On Wed, Feb 12, 2014 at 06:12:41PM +, Rodrigo Borges Pereira wrote:
Just one last question: do you have another suggestion about this?
thanks.
Not really. There are other ways that dahdi_cfg could serialize
itself, but POSIX semaphores are widely deployed on systems that are
installing
Ok thanks Shaun!
Best regards.
On Wed, Feb 12, 2014 at 6:22 PM, Shaun Ruffell sruff...@digium.com wrote:
On Wed, Feb 12, 2014 at 06:12:41PM +, Rodrigo Borges Pereira wrote:
Just one last question: do you have another suggestion about this?
thanks.
Not really. There are other ways
Hi List,
it feels silly, but here I am.
My Asterisk box is useless, after running a long delayed yum update (Centos
box).
*
A few details on the box:
cat /etc/redhat-release
CentOS release 5.10 (Final)
arch
i686
uname -a
Linux hermes 2.6.18-371.4.1.el5 #1 SMP
On Wed, Feb 12, 2014 at 10:44:42PM +0100, Aldo Bergamini wrote:
Hi List,
it feels silly, but here I am.
My Asterisk box is useless, after running a long delayed yum update (Centos
box).
[snip]
Starting Asterisk very verbosely seems to load the dialplan, but at some
point I get a
DAHDI might be the culprit.
You may have had a better version from Asterisk than the new one that
YUM got you.
Check to see if YUM gave you a new DAHDI. Who's your daddy now?
You may want to rebuild the Asterisk DAHDI and install it over the DAHDI
from your Linux distro.
Ron
On
Hi,
Is anyone aware of an issue with Gigaset DECT handsets (R630H and N510P) and
Asterisk?
A client has them, and whenever they try a blind transfer, something goes wrong.
Agent 1 starts and completes the blind transfer.
Agent 2 answers the transferring call.
Agent 2 hears asterisk music on
About a call not being hang up for asterisk while the client hang up,
please remember SIP is based on UDP and UDP packets get easily lost... they
are retransmitted but sometime they are lost as the previous...
For the ghost calls, are the SIP port of the phones reachable from the
Internet...
Ø when i use the Dial the sip/105 still ringing
This should help you out
http://wiki.snom.com/FAQ/How_to_make_Asterisk_send_INVITEs_to_trigger_the_phone_for_Intercom
Dan Journo
Kesher Communications (UK)
www.keshercommunications.comhttp://www.keshercommunications.com
--
In Asterisk 1.8, I used the following line in extensions.conf to allow
me to pass *82 in front of a dialed number, to disable the callerid
block that's normally on that POTS line:
; disable callerid block
exten = _*82.,1,Dial(${POTS}/${EXTEN})
But this seems to have stopped working when
On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote:
Hello,
How does extensions.lua compares to extensions.conf or extensions.ael on
stability, performance and features ?
Would you recommand extensions.lua as an easy/easier way to access
memcached, redis or equivalent ?
On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote:
Hello,
How does extensions.lua compares to extensions.conf or extensions.ael on
stability, performance and features ?
Would you
Hi,
I'm using SIP MESSAGE to asterisk V10 and it fails to be received.
I'm not super sure of the reason but I'm making this guess:
Due to I'm using non ipaddress in the to field, which contains
sip:mobil1.xyz.com, Asterisk makes the mistake to try matching this name
mobil1.testserver.com in
Since there is no Transfer button for SIP INVITEs, I guess it is a DTMF related problem. At
first I would check whether the ways of transmitting and receiving DTMF signals are compatible
(http://www.voip-info.org/wiki/view/Asterisk+DTMF).
I have a similar setup where there are no problems with
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