Hello,
If my understanding is correct, these days it seems that many ISDN BRI lines
are configured in energy saving mode in which signalling D-channel is
dropped until a new call comes in.
Is it possible to replicate this behaviour with Asterisk (when Asterisk is
in NT mode and is seen as a
Hello,
I'm looking for a virtualization technique with which I could easily assign
PCI/PCIe boards to virtual machines.
If this matters, I don't need to be able to use several boards nor to run
several virtual machines at the same time as I'm just looking for a way to
easily mimic several
Hi,
I'm looking for a virtualization technique with which I could easily
assign PCI/PCIe boards to virtual machines.
you might try opensource Xen with paravirtualized Linux-guests, which
supports PCI-passthrough quite good.
Running several Asterisk-servers at one time might be problematic,
2010/10/5 Andrew Latham lath...@gmail.com
http://www.ip-phone-forum.de/showthread.php?t=188877
For those not fluent in german, what is this thread telling ?
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more
On Wed, 6 Oct 2010, Olivier wrote:
Hello,
I'm looking for a virtualization technique with which I could easily assign
PCI/PCIe boards to virtual machines.
If this matters, I don't need to be able to use several boards nor to run
several virtual machines at the same time as I'm just looking
Thanks Steve,
I got the picture :) THANK!!!
But my doubt is about the cable, what cable should i use? i have a Sangoma
A108D in one machine (one machine with one card). What cable should i do?
how can i make it?
Best Regards!
2010/10/5 Steve Murphy m...@parsetree.com
On Tue, Oct 5,
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years. I
hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.
app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
app_mysql.c: In
Rizwan Hisham wrote:
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years.
I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.
app_mysql.c:33:25: error: mysql/mysql.h: No such file or
Hi Ridwan,
You would need to install mysql-devel via yum.
Best Regards,
Muhammad Nuzaihan Kamal
Network Consultant
Mobile: +65 97473874
Asfa Systems Pte Ltd
91, Alps Avenue. #03-10. Singapore 498787
Tel: +65 62538211
Fax: +65 62504814
www.asfasystems.com.sg
pub 4096R/36630777 2010-07-10
On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years. I
hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.
snip
Plz help.
You need
2010/10/5 Nikhil d.nik...@cem-solutions.net
Hi
We can run multiple instance of asterisk in same box with different IP
and port. U need to install asterisk in different location eg: 1:
/home/asterisk1/ 2 , : /home/asterisk2 ,and run both from that path ,
listen ip and port should be
Issuing the AMI Status command results in a list of active channels. But
how to figure out which channels are related before the call is
answered? 2 channels below are somehow associated, but how can I be 100%
sure they are related in order to implement a redirect of the incoming
call to another
Thank you all. It is now installed.
On Wed, Oct 6, 2010 at 5:04 PM, Steve Howes steve-li...@geekinter.netwrote:
On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years.
I hav'nt used it since 2008 (version 1.4.2). Now
The Loop Back Plug on the link you provided is correct. You take a
few inches of CAT5 and remove the outer jacket. Loop the wires into
the RJ-45 connector like the diagram shows and then crimp.
Ryan
On Tue, Oct 5, 2010 at 3:02 PM, Danny Dias ing.diasda...@gmail.com wrote:
Hello my friend
Hi All,
Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
versions.
Thanks
--
Best Regards
Rizwan Qureshi
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Hi,
if I Page more than one extension, then the MeetMe conference stays up
even if all the called extensions aren't available or are hung up.
Is there a way of keeping track of how many extensions are attached to
the conference, and require a number or a particular extension to be
present?
Hi James,
I have a smartNet as well, I contacted cisco for a login, so when I go
to the support section for 7900 series phones, then hit Download
Software, it lists this for the 7962G :
Expand all | Close all
Latest Releases
9.0(3)
All Releases
SIP v.9
9.0(3)
9.0(2)SR2
9.0(2)SR1
SIP v.8
8.5(4)
Hi list,
I was wondering if anyone had any solution to either one of two issues
I'm having:
I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware,
it works very well for the most part, but after less then a week of
heavy usage, eventually the phone gets into a state where it
I think you're right JR,
that has been my experience as well, I've run just great with sip
7.something firmwares for years.. unfortunately those don't support the
wireless headsets that the people I support want.
My boss really likes the cisco phones, the hardware feels really solid,
I wouldn't
Hi,
while testing current release candidate 1.8.0-rc2 I stumbled on a weird
behavior. I did not find any hints in the archives or at the bug
tracker.
Two SIP-Clients are connected (both on the local net, no NAT). The RTP
stream flows directly between the phones. If I set phone A on hold, the
On Wed, Oct 06, 2010 at 01:56:55PM +0200, Daniel Tryba wrote:
Issuing the AMI Status command results in a list of active channels. But
how to figure out which channels are related before the call is
answered?
CoreShowChannels gives a little bit of extra data in the originator
channel:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Wednesday, October 06, 2010 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Difference
Hi All,
Please can
Use Polycom, but if you really must use cisco phones, downgrade to 7.5.
I've got a lot of 79xx phones out there and 7.5 is the last stable release
as far as I'm concerned. It just seems to work, no periodic reboots
needed,
or any other quirkiness like with the newer firmware's. The feature set
It is describing the method of copying the German localization over
the United_States_English so the phones use German
~
Andrew lathama Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
Hello,
I would need a little help about using 16 bit wav or mp3 files for moh on
asterisk 1.2.x
When i try to use these files as moh, the caller gets disconnected.
Please advise.
Regards,
Z. Zivanovic
--
_
-- Bandwidth
Hi Danny,
We're using A104 cards which reduces the complexity of the necessary
cable (it's a regular T1/E1 cross cable).
In case of a A108, the cable that would come closest to what we're doing
is combining the A108 Straight Thru Y Cable and the A108 Cross-Over Y
Cable (or getting both and
Hi James,
I too facing the same issue whereas in the inbound call I am able to receive
the call, when I pickup the receiver it hangsup. I am getting the NOTIFY
option.. the log as follows,
-- SIP read from 98.158.181.173:5060:
NOTIFY sip:pbxfami...@10.0.8.84:5060 SIP/2.0
Via: SIP/2.0/UDP
Greetings,
I need a 2 way intercom for separate kitchens to communicate without
having to walk back and forth.
The speaker has to be loud but clear, not distorted. Sometimes the
kitchens can be noisy.
It needs to be easy to use.
It needs to be easy to clean.
It would be nice if it used POE.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 10:07 AM
To: asterisk-users@lists.digium.com
Cc: t...@casanueva.com
Subject: [asterisk-users] 2 way intercom
Is there any major architectural difference between 1.4 and 1.8?
On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan Hisham
Sent: Wednesday, October 06, 2010 7:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Difference
Hi All,
Please can anyone tell me
On Wed, 6 Oct 2010, Rizwan Hisham wrote:
Is there any major architectural difference between 1.4 and 1.8?
Nope. The developer's just got tired of typing .4
Of course, the joke's on them -- 1.8 is only .4 better than 1.4.
--
Thanks in advance,
Polycom 501's are pretty good and relatively inexpensive.
Danny,
Should I be worried that the Polycom 501 has been discontinued? What
does this even imply... that they won't be putting out any BIOS updates
(if there even is a BIOS on phones...)
Sounds like they'd be cheap to get on ebay
For a production environment, 1.4 is the most stable, and it has everything
one needs to setup a telecom platform. As per my understanding 1.6 never got
the same recognition for stability as 1.4, plus it doesn't have any
significant advantages over 1.4. The newer version 1.8 series might be my
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 way intercom
Back in the days i heard that they have changed the architecture in 1.6 and
its a lot better than 1.4 (6 times better call handling and robust
architecture, someone told me). If they have decided to take the 1.6
architecture to the next level in the new 1.8 version then its a good thing.
On Wed,
Here is a presentation from Kevin P. Fleming, Director of Software
Technologies at Digium. Information might be old by now still gives a good
overview of what is new in 1.6:
http://www.asterisk-tag.org/2008/slides/Kevin-Fleming-Asterisk-Tag-2008.pdf
Summary of his presentation is as follows:
–
I find 1.6.2.13 version is stable for trunk call routing, and it should
be too for basic call center use. The asterisk team has made some
architectural improvements (moving to astobj2 a lot of internal
structures, and much more you may not see from a user perspective) but
given the several
On Wed, 6 Oct 2010, Danny Nicholas wrote:
Polycom 501's are pretty good and relatively inexpensive.
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy
Graybeal
Should I be worried that the Polycom 501 has been discontinued? What
does this even imply... that they won't be
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http://www.eset.com
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On Tue, Oct 5, 2010 at 1:40 PM, Roger Burton West ro...@firedrake.orgwrote:
I now have an OpenVox A400P and it is working well. Thanks to Ade
Vickers for the recommendation, which I second.
However, I need to make a slow transition between a conventional
multiple-extension setup and a full
It is already a relative PITA to get BIOS updates - that being said,
when you are able to get them, there are plenty and Polycom is pretty
good about updating BIOS for discontinued phones.
It has been my experience that it is easier to get Polycom firmware
updates (just download off their
Is there a limit to the AMI connections?
I have upto 3 process that might connect and originate calls,
2 of those processes are very infrequent - however the main one
can issue a number of calls one right after the other.
I am seeing a message when I am logging failed manager connection
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 way intercom
Hey, all. While ADA can still be downloaded, that's about all that I see.
No development, no recent mention, and -- perhaps worst of all -- it
appears not to work properly under 64-bit systems. So, assuming Digium's
abandoned it, are there any suggestions of alternatives? Right now, I'm
Hi Guys,
This is such an annoying issue whenever it comes up. The sender and receive
always receive the source public IP no matter what in the IP packets but
then SIP packets go out with something like 192.168.0.20.
In this instance, a Bell Canada DSL modem is installed and I saw the
SPA-2102
In my experience, the 501 has very good speakerphone quality. It has 4
programmable buttons so the cooks can hit one button and connect. We have
one mounted on the wall in our computer room. Yes, the speaker is under the
handset, but you could take the handset off and tape down the switch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andy Graybeal
Sent: Wednesday, October 06, 2010 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 wayintercom
Polycom 501s were designed before the PoE standards were set in stone. So
the PoE is actually not part of the phone, but part of the special PoE cable
that is optional. So you absolutely need that special RJ45-like cable.
Mike
-Original Message-
From:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Wednesday, October 06, 2010 12:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AMI connection limit
Good afternoon list,
I'm having a problem using the function CALLERPRES() when connection to a
Queue().
When I call an extension, before the Dial (), I select the function
CALLERPRES () as unavailable to link the extension comes as anonymous. But
if I call a queue before the Queue (), I select
Hi list,
A few times I have been asked if I could do encrypted VoIP development, for
embedded systems, and in C++. And my answer has been in negative.
Now I am thinking I should start learning how to do it, but I have no clue
where to start from. I have been developing in Java for some time now,
Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone
system via a SIP trunk using the IPRC card?
--
Marvin Horst
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
On 09/22/2010 08:36 AM, Carlos Chavez wrote:
Do you have a localnet statement in your sip.conf? That and using
nat=no will make sure Asterisk does not replace the IP address in the
Invite.
I just wanted to give a +1 for this response. I am using openvpn to
connect road warriors and
On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
This is such an annoying issue whenever it comes up. The sender and receive
always receive the source public IP no matter what in the IP packets but
then SIP packets go out with something like 192.168.0.20.
In
Olivier wrote:
Hello,
If my understanding is correct, these days it seems that many ISDN BRI
lines are configured in energy saving mode in which signalling
D-channel is dropped until a new call comes in.
Is it possible to replicate this behaviour with Asterisk (when
Asterisk is in NT mode
On 5/10/10 8:17 AM, Alejandro Recarey wrote:
I just upgraded my asterisk box from 1.4 + Zaptel to 1.6 + DAHDI and
services I was using perfectly before are suddenly broken.
I have a DISA access configured, and my companies employees use if to
dial into the companies extension from their cell
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