On 19/6/17 4:47 pm, Tech Support wrote:
I know that there are probably several solutions to this problem, but
what I am trying to do is provide some redundancy for my customers CDR data.
I know that doing simple backups of MySQL is probably the easiest way to go,
but I'm thinking that there may
On 6/2/17 5:24 pm, Tech Support wrote:
Basically,
two calls are made.
...
When the first call is made for
such a short period, the remote end still goes off hook, but the call will
end before it starts to ring. Then, halfway through the first call, a second
call is made. Since the remote end
On 7 May 2015, at 23:45, Tafadzwa Nyabasa tnyab...@gmail.com wrote:
I am looking for a phone provisioning template for Snom phones, Yealinks and
Polycoms. I am always doing deployments of many phones and usually configure
each phone one by one for each installation. Any help will be highly
On 27/3/15 8:03 pm, James B. Byrne wrote:
One only accepts VOIP calls from known correspondents. I
am not clear why this is so other than vague warnings respecting
(admittedly real and serious) security issues.
Because on the whole most people don't *want* to receive calls from
random
On 16/2/15 4:13 pm, Andrew Colin wrote:
The strange thing is its only sometimes my dial string is as follows
exten = s,1, Dial (SIP/200,, tT)
For that particular route but obviously s,3 as have Ringing () first etc.
After she pushes ## 6 times it will go thru sometimes.
Are you sure it's a
On 29/10/14 12:59 pm, A J Stiles wrote:
Imagine what would have happened to the human race if Ugg the Caveman decided
not to share the secret of making fire with everyone freely, but instead went
around demanding shiny beads with menaces from anyone who just wanted to keep
themselves warm .
On 3/10/14 6:52 pm, Rainer Piper wrote:
the attacking server changed the destination Number at 18:53 CEST and
he is still blocked ... LOL
972597438354 callto:00972597438354
It's pretty much an everyday occurrence for any internet-connected SIP
system these days...
Oct 3 19:46:20
On 2/10/14 6:52 pm, motty cruz wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
${CALLERID(num):1} should do what you're after (or :2 if you need to
strip the + as well)
Kind regards,
Chris
--
This email is
On 24/9/14 10:36 am, A J Stiles wrote:
But personally, I'd just store the filenames in the database; and rely on the
unix filesystem for storing the actual file contents. After all, that's what a
filesystem is for.
This.
Shocking as it might appear, filesystems are remarkably good at storing
On 4/9/14 4:58 pm, Eric Wieling wrote:
If we don't need to allow access from outside the USA we block access from all
non-ARIN IP addresses by using iptables. This takes care of at least 80% of
attacks.
Likewise here (though RIPE rather than ARIN, since we're the other side
of the pond).
On 30/7/14 10:08 am, babak wrote:
I am evaluating some voice broadcasting solutions based on Asterisks for more
than 1000 simultaneous calls.
As a matter of curiosity, what do people use these voice broadcasting
solutions for?
I'm genuinely struggling to think of (legal) reasons why you'd
On 23/7/14 10:29 pm, Steve Edwards wrote:
Don't buy hardware until you've identified (either empirical or
calculated) the bottleneck.
If you've plenty of spare RAM (and at 16GB I'd suggest you probably do),
I'd throw in the possibility of recording to RAM disk, then moving the
calls to hard
On 20 May 2014, at 15:35, Ishfaq Malik i...@pack-net.co.uk wrote:
I was wondering if anyone has implemented voicemail to text and if so, what
package is being used to do so?
With the huge variety of different accents and intonations in human speech
(even in one country), my experience of all
On 24 Apr 2014, at 11:36, binary dreamer dreamer.bin...@gmail.com wrote:
I am running asterisk and all of my CDRs are in the default csv.
the system is so limited to ram (only 256) and I cannot run MySQL or any
other program to give CDRs a fancy view.
As an aside, have you considered running
On 22/4/14 11:44 am, A J Stiles wrote:
Firstly, be warned: Are you sure that is even legal to do in your
jurisdiction? You could be setting yourself up for a hefty fine! Check
applicable local laws before proceeding.
This. I'm glad someone else thought it worth mentioning as well :-)
Even
On 17/4/14 3:53 am, Lee, John (Sydney) wrote:
I have written a lot of AEL2 script in Asterisk 1.4.x and I am not sure if it
will still run in 11.
If I'm honest, this is why I still have so many 1.4.x boxes around as
well. I've been using 11 for new installs, but the thought of having to
On 17 Apr 2014, at 16:14, Paul Belanger paul.belan...@polybeacon.com wrote:
hi. I would not do that due to network issues.
My approach is to record everything locally and every hour or so to move
everything to a storage.
+1 save yourself the headache and do this.
I'll add another +1 to this.
On 17/4/14 4:53 pm, Eric Wieling wrote:
I had little problem converting my AEL scripts from 1.4 to 11
Did they have lots of macros in them?
If so, then you, sir, are a better man than I, and I take my hat off to
you :-)
(and any hints you might want to share in converting 1.4 AEL macros to
On 13/3/14 6:27 pm, Eric Wieling wrote:
This is an example of why I top post. Who wrote what?
Of course, if you use a mail client that's capable of quoting correctly,
it all works beautifully.
On 13/3/14 5:13 pm, Ron Wheeler wrote:
-1
Prefer top posting.
Easy to see if I want to scroll
On 6/3/14 3:21 pm, Thorolf Godawa wrote:
The idea would be having a HA-cluster of two servers with Xen, each of
them runs one instance of an Asterisk-system in a single VM and on a
failure the VM will be restarted on the other node.
This might result in a much higher load on this node, because
On 28/2/14 9:04 pm, Jayson Devor wrote:
That being said, will purchasing 23 licenses (one for
each channel that we use), and continue to use the open source g729
sorftware keep us legal?
I know at least half a dozen people who do this so that they can more
effectively balance their licence
On 19/2/14 4:53 am, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
I can't help on the can Asterisk detect they're behind the same
On 14/2/14 9:21 am, Gareth Blades wrote:
I would suggest using the 'M' option on the Dial command to run a macro.
The macro can just wait fir a key to be pressed and until it is pressed
the Dial is still effectively ringing. So if it does go to voicemail
then the call wont get put through. You
On 14/2/14 10:54 am, Tiago Geada wrote:
How does one detect the 'divert' to voicemail?
If you're using the mobile network's voicemail service, you can't as a
general rule; you've no reliable way of knowing whether that call was
answered by the user or their voicemail service.
However, if
On 25/1/14 5:26 am, Amit wrote:
How do I derive the requirement? I need to size IO system to record
multiple calls concurrently.
I suspect this might be your problem:
250GB SATA disk (No RAID)
Is there any way to tune / optimize / configure for better write
performance?
Perhaps consider
On 19/1/14 2:57 pm, Ron Wheeler wrote:
fail2ban is so easy to set up, there is no reason not to set it up.
One of the dangers with fail2ban - at least in its default configuration
- is that a legitimate SIP phone with an incorrect password can quite
easily send dozens of registration
On 10/1/14 8:16 pm, Jai Rangi wrote:
Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for thing really
good.
We recently used Ivona for a fairly complex IVR project (multi-lingual,
including pronunciation of
On 13 Nov 2013, at 18:29, Mike Diehl mdiehlena...@gmail.com wrote:
I've been seeing some strangeness lately on my 10.2.1 server. It's
gotten to the point that a few times each day, I see masses of SIP
clients becoming unreachable. They're not all on the same network,
and we don't see any
Is there a recent survey of that Linux distro and version people are
using for the Asterisk installations? I recall seeing a pie chart over a
year ago (I think on a wiki but I can't find it again)also hoping for
something more current.
Mix of Gentoo and Ubuntu here (Gentoo mostly on old
On 3/10/13 5:52 pm, Tech Support wrote:
I was
thinking of using sqlite3 to log CDR's, thinking that would be faster than
using MySQL. Has anyone ever benchmarked this to quantify just how much
faster sqlite3 is? Are there any drawbacks to using it?
Lack of multi-user concurrency is the big
On 20/8/13 5:00 pm, A J Stiles wrote:
Why not write an AGI script in your favourite language (Perl, Python, PHP,
Java all have AGI and MySQL bindings) to perform the INSERT query for you?
+1. It would also give you somewhere to perform sanity checks on your
${ARGS} to avoid SQL injection
FWIW, we routinely see dodgy traffic from:
ovh.net
hetzner.de
But since those are 2 of the larger short-term contract dedicated server
vendors, I'm not surprised about that. It's so frequent that I don't
even bother reporting it any more - when an abuse report is acted upon
and the server
On 5/8/13 2:18 pm, Jonas Kellens wrote:
is it possible to use the #include - syntax to include several
configuration files situated in one directory ?
Something like :
extensions.conf :
#include extra/*
#include addons/*
Is this possible ?
Yes.
You can also do crafty things like:
#include
On 15/7/13 3:00 am, bilal ghayyad wrote:
I need to be able to send SMS messages for campaign or for specific users, also
I need to be able to receive SMS messages and do automatic reply.
In my experience, SMS is something best done out of Asterisk. That's not
to say that Asterisk can't do
On 14/7/13 8:12 pm, bilal ghayyad wrote:
We have a cisco switches but they are not PoE and we need only to have PoE
device so the cables come for it first to provide the power and then goes to
the switch (to be like batch panel), is there something like this that can be
used for the IP
On 14/7/13 11:45 pm, Gregory Malsack wrote:
I've used a lot of Dlink DES-1228p and 1210-28p. Primarily with polycom phones.
Seem to have pretty good luck with them for the last 7 years or so.
+1. We've used quite a few DES-1228P units in the past, and apart from 2
early unit failures, we've
On 15/6/13 7:00 pm, Carlos Alvarez wrote:
Interesting product that I was very interested in, but the licensing has
one huge glaring problem. Be sure to read the FAQ carefully. If your
hardware fails and you replace almost anything in the machine, you have to
pay for the product again.
Not to
On 6/6/13 4:53 am, Gopalakrishnan N wrote:
Any other HA applications available or the lsyncd with pacemaker is good?
I generally use Pacemaker with Heartbeat, which seems to work pretty well.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
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On 2/6/13 2:01 pm, Muhammad Yousuf wrote:
I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm
gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have
g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1
and call leg from gsm gateway
On 22/5/13 10:54 am, A J Stiles wrote:
You do know that sort of thing is against the law -- or at least requires a
permit from the authorities -- in most civilised countries, right?
And it's worth adding that even if it is legal in your country, you're
almost guaranteed to offend/annoy your
On 21/5/13 4:19 pm, Ahmed Munir wrote: Last year, I installed Asterisk
10.4.2 and enabled logrotate on daily basis
which was working perfect. Now in couple of months back, the logrotate
feature is not working at all but simply appending the logs in 'messages'
file. Listing down down the
On 18/5/13 8:09 pm, Mitul Limbani wrote:
Not recommended to run Asterisk on Virualization
I used to share that view, but having done a few medium-sized installs
recently in virtualised environments and encountering no problems to
speak of, I'm not sure it's necessarily the case any more.
On 14/5/13 4:30 pm, Ishfaq Malik wrote:
I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my
primary goes down. I'm wondering what the best method of checking if the
primary being up is.
Well, the obvious start point might be ChanIsAvail() - that'll at least
weed out an upstream
On 26/4/13 10:38 am, jg wrote:
they are currently calling patients. I think these calls apply only to a
certain fraction of the patients, who are difficult to contact by other
methods.
I suspect there will be different requirements depending on how
'helpful' to patients you wish to be. At the
On 26/4/13 10:14 am, Hans Witvliet wrote:
Only reasonable option is to send them an SMS.
Given the likelihood that a sizeable percentage of people attending a
medical establishment are going to be at the upper end of the age scale,
it's possible they may not have mobile phones, and even if
On 26/4/13 12:24 pm, jg wrote:
This way the callees would always talk to a human being
If possible, this would definitely be a Good Thing. Many people (myself
included) will disconnect a call as soon as they realise it's a recorded
message. It also means the human caller can confirm they
On 12/4/13 4:38 pm, Nick Khamis wrote:
We were looking more into the lines of a
scalable multi server router like a cisco 3745.
Perhaps it might help to tell the list just how many concurrent calls
you're looking to transcode?
Kind regards,
Chris
--
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On 11/3/13 11:07 pm, Andrew Yager wrote:
Basically, if you know of a product, open or closed source, and would like to
sell it to me and you think it does the job, or you've seen something that
works, contact me off list ASAP!
Actually, please post *on* the list if you know or have used
On 7/3/13 6:50 am, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
I would caution against that approach. Analogue to Digital conversions
often seem to have 'problems' - mostly related to hangup detection
and/or echo. If you really do want to use analogue phones,
On 4/3/13 12:27 pm, Gertjan Baarda wrote:
After extensively googling the issue, I've found everything (also bug
related), accept my answer. What am I missing here?
It sounds like the call is being caught by a retry cycle on the queue.
Try adding n to your queue command from your dialplan.
On 18/2/13 5:39 pm, Administrator TOOTAI wrote:
on incoming call we have exten =
100,n,Dial(SIP/Handset_102SIP/Handset_103SIP/Handset_104,,)
and always only Handset_102 is ringing, we receive busy back from the
2 others but they are not. Any clue?
It depends which base station you're using -
On 17/2/13 5:02 pm, Administrator TOOTAI wrote:
customer102/Handset_102 xxx.yyy.zzz.153 D
N 5062 OK (80 ms)
customer103/Handset_103 xxx.yyy.zzz.153 D
N 5062 OK (70 ms)
customer104/Handset_104 xxx.yyy.zzz.153 D
On 8/2/13 12:11 pm, Doug Lytle wrote:
Is there a way to slow down or speed up the speed at which SayDigits
So, I'd have to say no.
I suppose potentially you could re-record the sound files to 'say' each
digit faster (and with shorter rolloff at the end of each word), then
put those into a
On 5/2/13 11:45 am, Jared Baxley wrote:
The closest building is 950 ft, the second is 1850 ft. These two buildings
are connected via LRE's using existing 6 Pair, Unfortunately re
cabling isn't an option. Other buildings are even further from the office,
about 15 or so scattered about that only
On 3/2/13 4:59 pm, David Smiley wrote:
I finally found the perfect solution for
me:http://www.amazon.com/La-Crosse-D111-101-E1-WGB-Wireless-Monitor/dp/B0081UR76G/ref=dp_ob_title_def
The device is $69, plus $10/month for alerts. And I get to monitor the
temperature online, which is a great
On 3/2/13 11:38 pm, bilal ghayyad wrote:
What should I do?
Given that you said:
This problem was not appearing when Asterisk machine was having static real IP
address because I was enabling the rtptimeout paramters.
I do believe the solution is simple: put it back on a public IP.
For what
On 1/2/13 12:41 pm, Luis H. Forchesatto wrote:
Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo
conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o
conector RJ45 está crimpada com a sequencia 568B e vai ser conectada na
placa Khomp, mas a ponta RJ11 eu
On 20/1/13 4:15 pm, Eric Wieling wrote:
Personally, I use the PHPAGI library and don't worry about all the low level
stuff.
This. It also gives you a nice logging function you can use to output
debug information to the asterisk CLI so you don't have to kill and
start asterisk interactively.
Greetings list,
I'm currently building a new cluster to replace our ageing Asterisk 1.4
infrastructure - it's easier to start from scratch then migrate users
across than it is to upgrade 1.4 to 1.8 in situ.
Anyway, it got me thinking about audio recordings in a multi-server
environment and
On 19/1/13 1:25 am, Joseph wrote:
I would like to outgoing/icoming calls and email the files.
This is what I have:
exten = _7.,n,Set(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = _7.,n,Monitor(wav,${CALLFILENAME},m)
How do I email these file?
You probably want to use MixMonitor() instead of
On 10 Jan 2013, at 22:09, C. Savinovich c.savinov...@itntelecom.com wrote:
Unfortunately, there is a fine line between being a forum where people can
exchange ideas, and being a forum where people can find asterisk consultants,
and both don't seem to co-exist well together.
Isn't this
On 29/11/12 6:33 pm, Dyweni - Asterisk-Users wrote:
I want to setup two Asterisk servers that are linked to each other:
- The first server would be my external (public) server and would live
in a real data center. The second server would be my internal
(private) server and would live in my
On 13/11/12 6:52 pm, Carlos Chavez wrote:
Why would a SIP to SIP call have this noise?
Check to see what random stuff they have on their desk.
We've regularly seen things like mobile phones (or cellphones to those
of you across the pond :-) ) causing interference with VoIP phones.
We've
On 13/11/12 9:31 pm, Leighton Brennan wrote:
It looks like you need to enable the sip application layer gateway or ALG on
your router
Quite often the reverse is true. Most routers (at least those I've used)
seem to have such a lousy implementation of a SIP ALG it's often far
better to just
On 4/11/12 8:37 pm, Danny Dias wrote:
For example, if i install a FreePBX/Elastix
I'd be very surprised (no, actually, I'd be *amazed*) if Digium were
prepared to provide support on a product from a third party, which is
what FreePBX and Elastix effectively are.
Kind regards,
Chris
--
On 31/10/12 6:20 pm, Darin Iv wrote:
is another way to build Multi Tenant system, have to design like
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
snip
Is there any particular reason why it needs to be _exactly_ like that?
FWIW, we use companyA-201,
On 25/10/12 9:49 pm, Justin Killen wrote:
What would be the advantage of using 100 single units vs. just buying VoIP
phones? That doesn't seem very cost effective to me in the long run.
In older buildings with existing single pair cabling, there might not be
a great deal of choice.
We
On 2/10/12 6:51 pm, Carlos Alvarez wrote:
Your traffic level, number of concurrent calls, etc would help us know what
sort of carrier you should be talking to.
Equally important, your geographic location, and the geographic location
to which most of your calls are made will be useful in
Greetings list,
I've seen a few errors recently in our logs along the lines of:
[Sep 10 17:41:41] WARNING[6719] app_voicemail.c: Save failed. Not
moving message: destination folder full.
maxmsg in voicemail.conf is set to 1000.
I've checked the mailboxes on the server in question, and the
On 10/9/12 6:48 pm, Danny Nicholas wrote:
What flavor of asterisk? Realtime or just files? Post your voicemail.conf.
Flat files, latest 1.4.x
Kind regards,
Chris
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--
_
--
On 23/8/12 5:26 pm, Adrian Marsh wrote:
I've a few questions around languages I'm on 1.4.18 (old yes I know, but
upgradings not an option just yet).
I've downloaded the gsm Japanese files from
ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place
I've found that when I switch to
Greetings list,
I have a scenario where half a dozen phones at a site appear to be
dropping offline for a few seconds every few hours, but the connection
between them and the asterisk server remains up.
It's been suggested to me that the problem might be to do with qualify -
which is
However, there's no reboot
button in the web GUI of the phone.
I have no experience with the phone in question and so will make no
comment regarding the OP's original problem, but the absence of a
software reboot function in the web GUI seems to be a pretty major
oversight in my view.
I do
On 26/7/12 11:08 am, Ishfaq Malik wrote:
I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...
We've tried Festival, Cepstral and Ivona.
Ivona was by far and away the best.
If you need free (or very low cost) then your only real
UK English is exactly what we're after. Did you try flite at all?
No, I wasn't aware of flite when we ran these tests.
Kind regards,
Chris
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-- Bandwidth and Colocation
On the subject of click to call - admittedly not necessarily what the OP
was after - I had some marketing blurb from VMware about Zimbra 8 this
morning. Apparently one of the new shiny features is integrated C2C (and
other unified comms stuff).
Has anyone had a chance to play with the SDK as
On 10/7/12 7:46 pm, Tim Nelson wrote:
Not to sound like a broken record or anything... but I'd say give Elastix a go.
It is top notch in terms of release quality and features. And, being based on
FreePBX, you can set it to 'Device and User' mode instead of the default
extensions mode so users
On 29/6/12 9:59 am, Ishfaq Malik wrote:
Does anyone have any experience of connecting SIP phones to an asterisk
server through the 2701HGV router that BT supply with their Infinity
product?
Good luck with that.
The BT 'Home Hub' and 'Business Hub' routers they supply with retail
ADSL and
On 29/6/12 11:16 pm, Michelle Dupuis wrote:
Can you really mix match any base station with any DECT handset?
Yes and no.
Do handsets have proprietary features which only work with their own
basestations? (eg: transfer between handsets)?
Yes. And that's the 'no' part of my answer above -
On 30/6/12 12:12 am, Michelle Dupuis wrote:
I like the look of the C610H. Is there a matching DECT base station by Gigaset?
I use the N300IP. Supports 3 active SIP calls I believe - and yes, does
have multiple SIP accounts (6, if I recall correctly).
Kind regards,
Chris
--
This email is
On 8/6/12 9:17 pm, Hiers, Richard wrote:
I don't expect to need to use any special hardware, just a sip trunk over our
broadband connection. We have about 150 phones at present. Is ESX a viable
platform for us? And second, what is the recommended virtual configuration
(mem, cpu, etc.)?
I've experienced this quite a few times, and after working with a local telco,
it has become policy to not place ADSL on lines where fax is going to be used
I too have seen this, and also with credit card processing machines in
shops that 'dial' the merchant bank to process transactions (in
On 9/4/12 3:04 am, Takehiro Matsushima wrote:
// I don't know what's difference t and T.
T allows the caller to transfer. t allows the called user to transfer.
You very rarely want Tt - since I doubt you want an incoming caller to
be able to transfer their call all over the place. You
On 15/3/12 3:45 pm, Jake Wicke wrote:
I'm wondering if any other Asterisk users have a recommendation for a reliable
SIP Trunk provider that supports Asterisk and offers decent support.
You should probably let the list know what region/country you're in, as
you'll want to be as close (i.e.
Greetings list,
I'm trying to source a very basic ISDN BRI - SIP gateway.
Unfortunately, everything I've seen seems to want to do lots of other
things - registering handsets, IVRs, voicemail, etc. I only want it to
present an ISDN BRI as a SIP account - I have an asterisk server for the
Greetings list,
I've done AGI scripting before, but in the past I've always wanted
control to be returned to the dialplan as soon as possible.
However, today I have a scenario where I want the script to remain
running during the playback of a file so that I can read DTMF at the end
of
On 22/2/12 2:55 pm, Danny Nicholas wrote:
You don't state the Asterisk version you are running, but personal
experience tells me you'd better invest in some Rogaine if you're depending
on the built-in stuff from AGI for DTMF input. I have personally wasted
weeks trying it.
Sorry, should have
On 22/2/12 2:50 pm, Zohair Raza wrote:
Try passing escape character
GET DATA $filename $timeout $max_digits $escape_character
Not sure I follow - according to the docs, there is no parameter
$escape_character
The problem seems to be that GET DATA returns control to the script
before the
On 22/2/12 3:39 pm, Ron Bergin wrote:
Have you tried increasing the timeout value in the command? To me, it
appears to be too short. Try setting it to 5.
Thanks for the suggestion - I tried 5, but no difference, it's not
waiting at _all_ for the playback to complete.
Just for
Greetings list,
Has anyone compiled (or could point me at) a list of the minimum required
modules and conf files for a very basic 1.8 deployment?
We have lots of 1.4 boxes in production, and I'm currently setting up a pair
of 1.8 boxes to bounce calls coming in via IAX over IPv6 over to the
Greetings list,
Whilst running through a routine check of some CDRs, I've noticed that the
originating channel's accountcode isn't preserved on creating a local
channel. For example, if we start with:
exten = 123,1,Set(CDR(accountcode)=foo)
exten = 123,n,Queue(bar,nrtw,,,)
And the queue 'bar'
Greetings list,
I've compiled and installed dahdi countless times on standalone machines,
but recently I've been trying to compile Dahdi in a Xen DomU without much
success. The errors I'm seeing are as follows:
ISP 10% rule is what you are asking about
expected that average usage is 10% of total subscribers with bursts
higher
But remember to plan well for those bursts and ensure you have sufficient
excess capacity. Certain events can have a significant effect on your burst
pattern: some
Am running 1.4.18 at the moment, and am trying to implement inline blind
transfer.
I have :
[featuremap]
blindxfer = *6 ; Blind transfer
Do remember that asterisk needs to be in the media stream for this to work,
so you'll want to make sure (in the case of SIP devices) you've set
I'm looking to build an Asterisk box that can run at a remote
location.
We've used the Asus eeeBox (desktop version of their little netbooks) quite
successfully in past projects: Atom 1.6, 1GB RAM, 160GB HDD.
Generally we run Gentoo Linux with Asterisk 1.4.latest, but no reason why
you
Actually, the Atom seems to be surprisingly powerful. We have a couple of
Atom boxes with transcoding and conferences enabled without issue. I
wouldn't pretend it'll cope with hundreds of conference participants, but
with ~10 or so it seems to be fine.
Likewise with transcoding - we've only
looking for your valued input on suitable suggestions for high quality VoIP
DECT
phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking
to a new manufacturer.
We've been using the Siemens Gigaset range for a few years now (specifically
C475IP and S685IP). Not had any
I was reading in the documentation about the SNOM phones (mainly 300)
but I did not find anything in the users-pdf's or on there
knowledgebase/website which would tell me if this is possible, there
is something for failover configuration but it is not explained at all.
It's highly appreciated
Anybody tried one with Asterisk yet ? Views ?
Apparently not available until the end of August.
We've certainly used the Snom 820 with Asterisk without any issue in the past,
and since both are based on (largely) the same software, I doubt there'll be
any major problems with asterisk
First things first. You are running /very/ old versions of firmware -
particularly on the 300 and 320. Upgrade them. I've been running
7.3.14 for some time without a problem, though it appears that 7.3.23 is
now out.
I concur about upgrading the software, but I'd stick with 7.3.14 for now
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