I am investigating High Availability solutions for my front end servers.
I got into a discussion regarding replicated local databases versus
a single fiber connected shared database on an EMC.
Is anyone running a shared database on a SAN? Care to comment on your
findings...
Thanks,
Robert
I use this all the time and am very pleased with the results...
sox ORIG_FILE.WAV -r 8000 -c 1 OUTPUT_FILE.gsm resample -ql
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday,
I have used Rhinos for a while and they are very stable and work well with
asterisk... You need a T1 port port though
I also just bought a Xorcom and that is working very well too... (This is USB
so no need for a hardware card)
-Original Message-
From:
I have 1 PRI and 1 EM Wink Circuit.
If I call a non working number and route it through the PRI, I get the
following:
RingingYou have reached a non-working number.
If I call a non working number and route it through the EM Wink
Circuit, I get the following:
NO RINGING... DEAD AIR
A
.
--
Zeeshan
On 2010-03-26 9:06 AM, Robert Grignon rgrig...@fleetone.com
wrote:
I have 1 PRI and 1 EM Wink Circuit.
If I call a non working number and route it through the PRI, I
get the following:
RingingYou have reached a non-working
Phone Assistance
Does a regular phone work on that port of the channel bank?
On Wed, Mar 17, 2010 at 5:00 PM, Robert Grignon rgrig...@fleetone.com wrote:
I have two Viking E10 Door phones and a Rhino FXS channel bank...
I have the channel set to immediate=yes and defined a custom context...
When
I have two Viking E10 Door phones and a Rhino FXS channel bank...
I have the channel set to immediate=yes and defined a custom context...
When I press the button on the door phone, the inside phone rings and I
can hear the person talk through the door phone... The problem is I cant
hear
I am very familiar Rhino Channel Banks and what to expect from them. I
am intrigued by the Xorcom USB Channel Banks simply because I don't have
to burn a hardware port... Can anyone comment on the Xorcom Astribank
(24 FXS channels) and how well it works in an asterisk environment?
I appreciate
Just curious to know how most of you deal with segmentation faults /
core dumps
These are a few of the things I have seen in regards to people dealing
with random crashes:
1. Apply Base OS updates
2. Recompile with DEBUG_THREADS and DON'T_OPTIMIZE turned on and look
for the cause of the seg
Come on, was that necessary? He was asking for help and considers it an
important issue... If you want to chastise the guy at least offer up a
solution for him...
Sam - I have never tried this solution but Sangoma has a reference to
this.
Discussion
Subject: Re: [asterisk-users] Can an agent Login to a queue and be
paused
I'm not sure if this works for newer versions of Asterisk, but on old
ones, you could pause an agent and THEN log him on, and he'd be paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought
versions of Asterisk, but
on old ones, you could pause an agent and THEN log him on, and he'd be
paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought there was an option for this but cant find
it
We
I thought there was an option for this but cant find it
We have a busy callcenter and I would like the agents to log in and be
in a paused state upon login... Right now they login and they are
instantly receiving a call
Thanks for the input...
--
I have asterisk 1.6.1.10 and a Rhino CB24 Channel Bank...
A few channels seem to have locked up... If I plug an analog phone in
the port, I get either dead air or a busy tone...
Is there any way to reset this channel without restarting asterisk?
___
Sorry if this is off topic
I have a loud talker in our call center and was asked if I can make
his voice louder to make him talk softer :-)
Does anyone know if you can do that with Polycom 430's
I found voice.gain.tx.headset but wasn't sure if that will make his
voice louder to the calling
I was dealing with an issue for a few weeks with my Gateway randomly
crashing (Didn't matter what version of asterisk, sangoma firmware,
etc)... I finally hooked up a modem cable to serial console and was able
to catch the crash.. Wanpipe was causing it
I spoke with Sangoma and they said
What Hardware are you using?
What OS are you running?
If your getting a kernel panic you can install a crashkernel (kdump) and
upon receiving a kernel panic it will reboot to a crashkernel, capture
the crashinfo and safely reboot the system. You can then use the crash
utility to analyse the
This can also be caused by IRQ conflicts. You could try a different slot
to see if it clears it up
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir
Cohen
Sent: Thursday, November 12, 2009 1:44 PM
To:
I am about out of ideas
I am not able to keep this gateway stable. I am crashing about 2 times a
day
Is there a way to capture the crash data? I have kdump configured on the
server but it seems to be a hard lockup and not a kernel panic
I have tried the following:
Asterisk
Testing a new gateway and have a Rhino Channel Bank... Sending a test
fax and everything works fine (Receive the fax fine) But I notice this
in the log
Google search didn't return much of anything...
DAHDI hook failed returned -1 (trying 1): Device or resource busy
www.voiceroute.org also has an open source unified communications
manager (they also have a commercial version)... Very little support
from the developers but I have deployed it in a few large call centers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I have a PRI and a LDT1 (em) running...
When placing a call through the PRI (to a number with an auto
attendant). I hear thank you for calling. Please press a number
When placing a call through the LDT1 to the same number. I hear
...Please press a number
It is cutting off the Thank you for
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Grignon
Sent: Thursday, October 29, 2009 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Delayed answer when calling out
I have a PRI and a LDT1 (em) running...
When placing a call through
This has been a rollercoaster ride
Building a new gateway (Asterisk 1.6.1 / Sangoma A108D 3.5.8 drivers)
Where I stand right now, I have a PRI on the gateway and circuit is
working I can make calls through the gateway
Here is my problem:
DAHDI_TEST is not returning anything and
] On Behalf Of BJ Weschke
Sent: Wednesday, October 28, 2009 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Having a heck of a time
On Wed, Oct 28, 2009 at 2:09 PM, Robert Grignon rgrig...@fleetone.com wrote:
This has been a rollercoaster ride
Building
Yes I did that...
I even recompiled dahdi-linux and tools after wanpipe install... Once I
did that it recognized the card and said I could run dahdi_genconf
modules which in turn would only load the cards that it seeing.
I had the PRI running in slot 6. Once I unplugged the PRI I was able to
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Grignon
Sent: Wednesday, October 28, 2009 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Having a heck of a time
Yes I did that...
I even recompiled dahdi-linux and tools
We have queuemetrics and it does that
Here is some of the logic - (Obviously this wont work for you right out
of the box but you should be able to decipher the logic...)
[qm-queuedial]
; We use a global variable to pass values back from the answer-detect
macro.
; STATUS = U unanswered
;
I did run into some issues with this as well. I ended up setting
format=wav and left it at that... It wasn't so much a problem with
someone leaving a message rather when someone was forwarding messages. I
would have used wav49 but people were having problems getting wav49 to
open on their PDA's
- not leaving the
message but forwarding it (I suppose the correct term rather than
transfer). Thanks - John
On Thu, 2009-10-22 at 10:29 -0500, Robert Grignon wrote:
I did run into some issues with this as well. I ended up setting
format=wav and left it at that... It wasn't so much a problem with
someone
Just looking for some ideas here...
Single office with 1.4.26.2 - Frontend 1.4.26.2 w/sangoma A108 Gateway
I have been getting a few complaints about caller cant hear me or I
cant hear the caller I've listened to the recordings and can verify
what they are complaining about, with this being
I wont say we are extremely high volume (40 concurrent calls) but I get
occasional complaints about quality.
Setup (at same location):
Asterisk 1.4.26.2 FrontEnd
Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1
Connected via IAX2 trunking on its own VLAN
Is IAX2 the way
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