Any 2000+ user Asterisk PBX installs out there?
Please hit me off-list, I need some support on a 2000+ user Asterisk PBX
with high availability and over 10E1s to PTOs
Femi
I would be interested in some of the replies if you wanted to continue the
topic on-list... Your problem might help
Hi,
I want to estimate the amount of bandwidth required for Asterisk running
on
a T1 in a typical scenario.
Can someone share with me any implementation experience?
Thanks in advance for your input.
Regards,
Mark
Check out http://www.asteriskguru.com/tools/bandwidth_calculator.php it
Hello,
I wonder how Asterisk scales when we increment the Core's or CPU's of
one computer.
I see that Asterisk is only one process (I guess that it uses threads).
But because Asterisk is only one process, this process is always
executed in the same CPU. So we can have a 8 Cores server,
At 11:53 AM 1/18/2008, you wrote:
Apart from the fact asterisk 1.2 is in security maintenance
mode only and wont get any other bugfixes it will be ok.
Please consider using 1.4 as it's the official latest stable
version.
Although for some of us, or at least me, no version of 1.4 has run
for
Tilghman Lesher wrote:
On Friday 07 December 2007 20:12:12 Philip Prindeville wrote:
Darryl Dunkin wrote:
You can store most of the configurations in a database which may be
more
accessable to you.
Perl can also parse these configurations quickly enough if you know
how
to use the input
Hi.
I wanted to write a popcorn app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten = s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten = s,n,While(${EPOCH}
In article
[EMAIL PROTECTED],
Ryan Burke [EMAIL PROTECTED] wrote:
I just was looking over the app_waitutil.c and am confused you add 500
to
tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 -
((tv.tv_usec
+ 500) / 1000);?
It's just doing a standard round to nearest integer
,
Alex
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Burke, A+, CCNA
Kingston, Ontario, Canada
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it entirely within the dialplan.
Any ideas?
On 10/27/06, Alexander Burke [EMAIL PROTECTED] wrote:
Hello, all!
I'm having a problem with the following snippet that executes upon hangup:
exten = h,n,Wait(5)
exten = h,n,System(mv /some/file /some/other/dir/)
Wait() doesn't want to seem to wait! So
/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
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I don't have access to the sip code right now, but from past network code
I've writen you could try this:
inet_aton(192.168.1.10, p-sin.sin_addr);
If they are just wrapping the struct sockaddr_in as sin in p. Worth
a try...
Ryan
Hello list, I am trying to include a new message after I
Do me a favor and try running netstat -aplntu | grep asterisk and see
what ports are actually being used. Are you connected to another ITSP? If
so then that may be the local port of that connection... just an idea, i
don't have Asterisk access right now to double check.
Ryan
On 10/16/06, Time
Sorry, just checking if my mail is working.
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I would like to know how you got Asterisk to function with 2500 SIP
registrations. Did you have qualify enabled?
Yes, qualify was enabled, using the standard length of qualification
period between checks. Very few accounts had custom qualify settings.
What about the 500
Benjamin Jacob wrote:
Rushowr wrote:
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- Original Message -
From: Rushowr [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 19, 2006 10:38 AM
Subject: Re: [asterisk-users] When does Scalability requests Asterisk to Use
SER ?
Thanks for
I had that running in my small home Asterisk setup for a while, however I
had to move back to an older kernel to support a SATA driver (mv_sata). It
looks like as of late 2.6.16 and 2.6.17 they've updated the driver so in a
week or so I am going to try builind it again and create a DomU with
bailing before I have a chance
to clean up the recording in priority 10?
Thanks in advance!
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bailing before I have a chance
to clean up the recording?
Thanks in advance!
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Kingston, Ontario, Canada
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Just in case anyone here hadn't noticed, Cisco is apparently making
7940/7960 SIP 8.2 firmware freely downloadable by anyone:
http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960
username: anonymous
password: your email address
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Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
Sounds great.
I'm just a home user of Asterisk, but I love the
product and have recommended it to alot of other people. Let us know when the
site is up.
Ryan
- Original Message -
From:
Bruce Reeves
To: asterisk-users@lists.digium.com
Sent: Thursday, April 13,
I'm interested but I'm in the Dallas area. Are
there any in the Dallas area anyone knows of?
Ryan
- Original Message -
From:
Bruce Reeves
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, April 10, 2006 12:51
PM
Subject: [Asterisk-Users]
done, too. :)
Thanks for the laugh, Andrew!
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was released for the
7940/7960? Has anyone tried it yet? If so, what are people's opinions?
Thanks in advance!
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Kingston, Ontario, Canada
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(which never comes), or even
before a call-alerting event. This is also the case where a SIT is
generated, and a message like the number you have reached is not in
service is played for those not hardcore enough to know the specific
error from the sound of the SIT alone. :)
--
Alexander Burke
it, though!
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Kingston, Ontario, Canada
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?
Thanks in advance!
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Kingston, Ontario, Canada
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another way to power your device.
Sorry!
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Kingston, Ontario, Canada
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At 05:03 PM 02/26/2006, you wrote:
I want to match the user from the users callerid. All users have DIDs.
You probably shouldn't do that for security reasons -- rather, match
them according to the SIP username/password pair they provide when
they register.
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Alexander Burke, A+, CCNA
ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy
Remove the hash* (#) symbol from in front of
ztdummy, save the file, and compile
Zaptel as usual.
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/servers/entry/x2100/
These boxes will run Solaris, Linux, or (ack) Windows, and their
remote monitoring/management support is second to none.
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to do some work.
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Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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SysV-based like Linux, it
shouldn't be too much of a stretch.
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Kingston, Ontario, Canada
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and conferencing in a real setup.
Thanks again!
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Kingston, Ontario, Canada
Date: Sun, 19 Feb 2006 23:45:01 -0500
To: asterisk-users@lists.digium.com
From: Alexander Burke [EMAIL PROTECTED]
Subject: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
Hello, world!
I'm
, too!
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sharing motherboard IRQ resources.
Regardless, it is still worth considering
using multiple servers, as the redundancy that can be gained from
this strategy can quickly offset the cost.
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Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
it was available online, I
still would have bought it on paper.
You're welcome to download it and keep it on hand -- it makes for
EXCELLENT reading:
http://www.alexburke.ca/asterisk-tfot.pdf
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Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
(or lack thereof) that you have with it!
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Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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:55:50 -0500
To: asterisk-users@lists.digium.com
From: Alexander Burke [EMAIL PROTECTED]
Subject: Download Asterisk: The Future Of Telephony
Hello, list!
I'm hosting a mirror of the book Asterisk: The Future Of Telephony
by O'Reilly Press, published under the Creative Commons license; I
believe
(including music-on-hold and conferencing)? If so, how?
Thanks in advance!
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Kingston, Ontario, Canada
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List,
I purchased an SPA-1001 from voipsupply.com (which
was great.. no hassle), but now anytime there is silence and then Playback()
sound files the first half second gets cut off. For example, when I login to the
VoiceMailMain() function I don't hear "Password",I hear "-assword". Then
Any ideas? I appreciate hte help.
Ryan
- Original Message -
From:
Ryan
Burke
To: asterisk-users@lists.digium.com
Sent: Wednesday, January 25, 2006 7:58
PM
Subject: [Asterisk-Users] Speech playback
getting cut off
List,
I purchased an SPA-1001 from
Sorry, I haven't received a message in a few hours, just testing to see if
it is alive.
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Buffers don't have to be in memory. My suggestion on the solution would be
to buffer the CDR info into a backup file based database (configurable
filename/path) on the local filesystem (or NFS mounted system for
redundancy) and then when the SQL database connection is restored then it
spends a
hi,
i have an issue that when making a call from a SIP phone going as follows:
phone -- asterisk -- cisco(192.168.0.1) -- terminating voip
platform(10.0.0.1)
i get the cisco sending up an invite to the voip platform followed
directly with a CANCEL message, as follows:
Via: SIP/2.0/UDP
hi,
i would like all my calls originating from asterisk users bound for
external to route to one destination, a session border controller.
protocol used is sip.
i have edited extensions_custom.conf with:
exten = _.,1,dial(sip/[EMAIL PROTECTED])
would this be correct to send any calls from
Asterisk-Users,
Is anyone out there using the SPA-1001 for integrating existing analog
phones into a VoIP setup? My question has to do with the MWI. From the
datasheet it says that it provides MWI Tones, and then that it provides
Visual MWL via FSK. What does via FSK mean? My exsting phone has an
I'm currently using CentOS 4.2 in my home install on a P3-600/512MB/40GB
HDD with a X100P clone and it works great. Using Asterisk 1.2.1.
Ryan
Any thoughts on CentOS-4.2?
It is based on RHEL4 update2.
It has the 2.6 Kernel.
I am currently using CentOS-3.5, which is based on RHEL3 update5,
I had the same problem at first. Try adding a w or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
Hi all :
I need a little help please.
I have
Teliax users,
I have a couple questions about Teliax, just
hopeing some current customers might shed some light on them.
How reliable is a toll-free number from Teliax? Has
anyone had any problems with it?
The Pay as you go plan has a Billing of 60/1, what
does that mean? My guess is 60
]
To: asterisk-users@lists.digium.com
Sent: Saturday, December 17, 2005 7:05 PM
Subject: [Asterisk-Users] Re: Teliax billing question
Ryan Burke [EMAIL PROTECTED] writes:
Is there any other charges because of the toll free number?
I was toying with the idea of getting an 800 number too, but the issue
Rich,
Thanks for your feedback. Sounds like what I was looking for. I think I'll
sign up tonight!
Thanks,
Ryan
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday,
I was reading the pdf and found a command that might be of some use:
Prefix()
ex.
exten = 8661234567,1,Prefix(1)
exten = 18661234567,1,NoOp()
exten = 18661234567,2,Goto(800-in)
After the Prefix() the the next exten is n+1 (which is 2 in this example)
with the new extension (which is
Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes. One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between
WEll I personally have not implemented a Linux-HA cluster mainly because I
don't have the resources to do so. I study Asterisk purley as a hobby
(nerd.. yeahI know) because it is an awesome OSS product. Anyways, after
some searching around I think it would not be TOO difficult to implement a
seconds
I realize that it's a timeout but what's implicit in that is that
Asterisk can't detect # of rings just the amount of time spent ringing?
I have been looking at the reference manual on asteriskguru.com. They
say it's a timeout but they don't indicate the units. Is it
milliseconds,
This sounds like a prime candidate for a database implementation. That way
you can get very near real-time stats without the overhead of frequent
cronjobs or polling. You number crunching computer would then just grab
the data and crunch away. I'm just now getting started on using Asterisk
in the
Here is an example, very basic.
sip.conf
=
[1000];assuming that your first SIP extension is 1000
... ;all the other paramters
context=internal ;default context is [internal]
extensions.conf
[globals]
OUTBOUNDCHANNEL=Zap/2
[internal]
include =
While I'm not sure about the 1.2 ChangeLog the 1.2.1 that was released
recently has an real Changelog that is linked at the top of Asterisk.org.
http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.1
It looks like they put some effort into this one so I hope it continues for
future releases.
On Mon, 21 Nov 2005 18:43:34 +0100, Josh wrote:
Hi all,
I'm kinda new with asterisk stuff.
I'm running a Debian with asterisk and a digium X101P clone card in
country #1.
Since I'm going to work in another country (country #2), I would like
to setup another Asterisk server + 1 FXO device in #2
Yep, I second (or third) that observation.
Ryan
It's not just him. The list was majorly down from sometime on the 29th
until the 1st.
MARK.
Derek Whitten wrote:
must be just you.. get messages all day every day here..
:-)
On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote:
This
Yes I would be very interested to see some kind of example if anyone has one
running?
Thanks
- Original Message -
From: Paul [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, April 01, 2005 11:19 AM
Subject:
Hello everyone, I've been watching this list for a while, but it is the
first time I've posted. I'ved decided to setup a * server for my house and
will need 3 phones (one main, one for my wife, and one for my office). I was
wondering if there was a particular brand that people reommended? I'd
running my existing phoes through an
adapter. Thanks for the suggestion though, it is definately good to know and
a possibility in the future so that I can use some of my wireless phones
with *.
I'm always ready for more suggestions though..
Ryan
- Original Message -
From: Ryan Burke
: [Asterisk-Users] Recommended Phone for beginner
Ryan Burke wrote:
Wow.. what an awesome mailing-list!
I appreciate the input, I'm looking at the Polycom 300 right now and
debating on spending the extra $60 / phone to upgrade from the BudgeTone.
Chris, I like hte Sipura 2100 idea, but I need
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