> 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
> 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_calculato
>
> 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 se
> 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 h
> 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 y
> 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
&g
> 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
,
Alex
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Kingston, Ontario, Canada
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ion about doing 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/di
/var/spool/asterisk/outgoing/)
This only executes sleep, not mv. How can I make it wait before
moving the file?
Thanks in advance!
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ast
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 afte
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, Tim
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
- Original Message -
From: "Rushowr" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, September 19, 2006 10:38 AM
Subject: Re: [asterisk-users] When does Scalability requests Asterisk to Use
SER ?
Thanks for the info. So it was reall
> Benjamin Jacob wrote:
>> Rushowr wrote:
>>
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>
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
As
#x27;
How can I prevent the extension from bailing before I have a chance
to clean up the recording in priority 10?
Thanks in advance!
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Kingston, Ontario, Canada
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#x27;
How can I prevent the extension from 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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Asterisk-User
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
--
Alexander Burke, A+, CCNA
Kingston, Ontario, C
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, 20
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]
10, SIP 8.2 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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As
uot;419" scam email. Rather well done, too. :)
Thanks for the laugh, Andrew!
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Kingston, Ontario, Canada
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event (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. :)
--
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.
--
Alexander Burke, A+,
find another way to power your device.
Sorry!
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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plications?
Thanks in advance!
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Kingston, Ontario, Canada
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od things about it, though!
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ctually have to do some work.
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any PCI cards):
http://www.sun.com/servers/entry/x2100/
These boxes will run Solaris, Linux, or (ack) Windows, and their
remote monitoring/management support is second to none.
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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--B
wcusb wcfxo wctdm \
ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy
Remove the hash* (#) symbol from in front of
ztdummy, save the file, and compile
Zaptel as usual.
--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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--Bandwi
: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 u
ake, and the success (or lack thereof) that you have with it!
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Alexander Burke, A+, CCNA
Kingston, Ontario, Canada
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online,
but it's good enough that even had I known 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
sets, and thus can handle 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.
--
Alexander Burke, A+, CCNA
eilly for setting it free, too!
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e people going without MoH and conferencing in a "real" setup.
Thanks again!
--
Alexander Burke, A+, CCNA
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 (AM
like to learn Solaris, and being SysV-based like Linux, it
shouldn't be too much of a stretch.
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k
solution on Solaris 10 (including music-on-hold and conferencing)? If so, how?
Thanks in advance!
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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
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
a
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 seco
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 1
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 update
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
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
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.
>
>
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"
Sent: Saturday, December 17, 2005 8:30 PM
Sub
IL PROTECTED]>
To:
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 to
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
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 186612
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
> millisecon
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
resour
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 th
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 =>
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 m
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.
Rya
> 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 F
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, H
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'"
Sent: Friday, April 01, 2005 11:19 AM
Subject: [Asterisk-Users] Is this pos
sterisk-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 ne
es vs 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: &
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 li
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