On Wed, Jun 17, 2015 at 10:01 PM, Mordechay Kaganer mkaga...@gmail.com
wrote:
B.H.
Hello, all.
We have noticed many calls on our PBX get stuck - the other end sends
BYE, and our side sends ACK but the call remains active (no hangup event on
AMI, the call is listed in 'core show channels
B.H.
Hello, all.
We have noticed many calls on our PBX get stuck - the other end sends
BYE, and our side sends ACK but the call remains active (no hangup event on
AMI, the call is listed in 'core show channels') and it's impossible to
hang up until asterisk is restarted. Asterisk's log shows
B.H.
Hello, all :-)
We have a cluster of Asterisk (v. 11.9) servers that host IVR applications.
The servers work behind SIP proxy (kamailio) for load balancing.
All servers are in 2 processor configuration, 8-10 cores per CPU.
When a particular server gets about 500 concurrent calls, the sound
B.H.
Hi, really thanks for all the relies :-)
Here's my answers:
On Mon, Mar 2, 2015 at 5:56 PM, Ron Wheeler rwhee...@artifact-software.com
wrote:
Have you done the math for the network connections? BTF and external
What bit rates for the sound?
What codecs?
How are calls coming in - SIP
B.H.
Hello, all
I'm using Asterisk 11.7, connected to PSTN using SIP trunk.
I'm looking for a way to get data from INVITE's SDP. Specifically, i would
like to get a value of o= for incoming call from PSTN because it contains
data about the operator that the call originates from.
I have googled
B.H.
Thanks for your response
On Aug 23, 2013 11:21 AM, Gareth Blades mailinglist+aster...@dns99.co.uk
wrote:
On 22/08/13 15:43, Mordechay Kaganer wrote:
B.H.
Hello, i'm using AMI Originate action (with async=true) to send outgoing
calls to a SIP trunk (using asterisk-java library
B.H.
Hello, i'm using AMI Originate action (with async=true) to send outgoing
calls to a SIP trunk (using asterisk-java library to connect to AMI).
The problem is that in case of failed originate, OriginateResponse event is
returning only the reason code which is sometimes not sufficient to
in the previous mails, SIP response code is 480. I would
expect to get reason 3 not 8. Reason 8 is confusing my dialer software so
it wants to redial the number.
I use Asterisk 1.8.22. Is this a bug in asterisk or is a problem with my
SIP trunk provider?
On Wed, Aug 14, 2013 at 9:00 AM, Mordechay Kaganer
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mordechay Kaganer
*Sent:* Tuesday, August 13, 2013 10:55 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] SIP trunk and congestion handling
** **
B.H.
Asterisk
B.H.
Asterisk 1.8.22
Thanks
On Aug 12, 2013 8:05 PM, Shishir Pokharel shishir.pokha...@on24.com
wrote:
Which version of asterisk are you using ?
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mordechay
B.H.
Hello, all. We have a dialer software that runs outgoing telephony
campaigns. We have been using it successfully with PRI cards, now we're
evaluating it's use also with a SIP trunk. Most of the things run perfectly
good without a need to change anything except for dial string, but there's
B.H.
On Wed, Jun 19, 2013 at 10:20 AM, Grant Bagdasarian g...@cm.nl wrote:
Hello,
** **
I’d like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
** **
Whenever I execute the below action, the recipient does
B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.
In past we were using asterisk 1.4 on the server that is receiving IAX
connections and
B.H.
On Tue, Jun 11, 2013 at 3:45 PM, Doug Lytle supp...@drdos.info wrote:
WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
from address X.X.X.X
I don't know if this will help, but I have:
requirecalltoken=no
In my iax.conf
Doug
Thanks, Doug. I too have it there
B.H.
On Jun 11, 2013 5:15 PM, Steve Totaro stot...@totarotechnologies.com
wrote:
On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.com
wrote:
B.H.
Hello!
We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2
B.H.
Hi!
On Wed, Jun 5, 2013 at 7:26 PM, jg webaccou...@jgoettgens.de wrote:
For a BRI device a single span has 2 channels, a PRI device up to 30. As
far as channel variables go the actual channel does not seem to get
reported, but this is not really necessary.
AFAIK, at least for AMI
B.H.
Hello, all :-)
We have some OPENVOX D410P PRI cards and we are successfully using them
with Asterisk boxes which are based on stock ubuntu 12.04 DAHDI and
Asterisk packages.
The card is recognized by DAHDI as 'Wildcard TE410P (2nd Gen)' and it uses
wct4xxp driver.
Now, i'm trying to run
B.H.
Thanks for the reply
On Mon, Jun 3, 2013 at 7:42 PM, Russ Meyerriecks rmeyerrie...@digium.comwrote:
On Mon, Jun 03, 2013 at 06:34:42PM +0300, Mordechay Kaganer wrote:
The card is recognized by DAHDI as 'Wildcard TE410P (2nd Gen)' and it
uses
wct4xxp driver.
What's the output
B.H.
On Mon, Jun 3, 2013 at 9:15 PM, Emiliano Vazquez
emilianovazq...@gmail.comwrote:
Openvox have anothers modules to compile dahdi.
You need compile with the latest openvox-dahdi [1] or diff this with the
latest dahdi from digium.
Thanks! The patched version worked 100%. Don't ask me why
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