isconnect
cadences...
--
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nr k wrote:
> hi generally we describe the bandwidth in kilobits per
> second only.
Cool, just checking, it seemed pretty low.
According to http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 you should
be able to do 4 calls with g729.
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allation, as they were mainly done in a lab condition for a lab
> test.
We're running supermicro rack mounts (1U) in high load production environments
and have been nothing but impressed.
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Colin Anderson wrote:
> Onboard LAN with an un-movable IRQ would mess that up good
Only if you had just one pci slot.
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Jonathan k. Creasy wrote:
> I guess I should have read up further before I posted a response.
Nah, it's all good. If multiple people post replies to questions, then it
helps to confirm the truth of a statement!
:D
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_
Kevin P. Fleming wrote:
> Kanuri, Seshu (Company IT) wrote:
>
>> I totally agree. But doe the Asterisk list servers have any such feature
>> to block the spam and delete the spamming users? I don't think so.
>
> It's been taken care of.
Thank
gh some other way. The threaded
emails are perfect for me.
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his information applies to both the
> Intel Desktop Board and Server Board product lines.
So just install Digium cards and disable anything you don't need. You only
need to change IRQs if you have an unresolvable conflict.
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looking at cadences rater than frequencies.
Do you need tone detection for hangup detection or do you get polarity
reversals? If polarity, you might be able to turn off busydetect and just use
the reversals.
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, how do I make this
> phone recognize that there is a voicemail waiting, and flash it`s red led?
You need to set the mailbox for your FXS channel IIRC to enable stutter
dialtone.
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xt
Usage: RECEIVE TEXT
Receives a string of text on a channel. Specify timeout to be the maximum time
to wait for input in milliseconds, or 0 for infinite. Most channels do not
support the reception of text. Returns -1 for failure or 1 for success, and
the string in paren
ts back to the bug tracker
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ranch i am using g729
> codec(for testing).
is the 64 kilo bits or kilo bytes?
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ssful
-- Attempting native bridge of Zap/12-1 and Zap/25-1
-- Native bridge of Zap/12-1 and Zap/25-1 was unsuccessful
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Mark Phillips wrote:
> So I guess we won't be seeing the fruits of your labours then?
Not yet - it's a real hack - all done in php because I couldn't compile c
programs on this host.
Once I get a spare moment I will try and write a generic one.
--
Che
chawki hammoud wrote:
> Hi:
>
> I posted my problem several times about being unable
> to make IAX calls from my Asterisk box to another IAX
> server without luck.
So, what's your problem?
Post some details.
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_
ree open later that week."
So, yes it is available in 1.2, but it is not in the current STABLE - 1.0.9
(at least until early next week - when 1.2 becomes the new STABLE version).
:D
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Bartosz Piec wrote:
> Matt Riddell napisał(a):
>
>> You cant do fax in g723.
>
>
> So what to do? Change the fax machine?
:) No, make sure that whatever it is connected to is using ulaw for its audio
codec.
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eds an RS232 connection and mine is dead at the moment.
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I'd really reccomend reading the Asterisk Book:
http://www.sineapps.com/news.php?rssid=1044
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Fabio Montemaggiore wrote:
> Why the estension s dont' start?
Do you get an error in the Asterisk console?
A good thing to read is the Asterisk Book which you can download for free from
one of the mirrors provided here:
http://www.sineapps.com/news.php?rssid=1044
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away if you have questions!
:D
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harry gaillac wrote:
> What about egroupware !
We use it, although there is no simple click to install installation package
for Asterisk integration.
The idea is to use flash operator panel to load a url when each extension is
dialed. And for click to dial, I use call files.
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to make a call using IAX, the call get
> accepted and then get a hangup message:
is this the same number format you send when using sip: 0017046872001
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he announcement here (
http://www.voip-forum.com/news.php?p=183 ), but it doesn't really have that
much info.
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gt;
> In the xlite configuration, look for an option something like 'transmit
> silence' and set that to yes. (Might be called 'silence suppression', I
> don't remember.)
I think it's called VAD (Voice Activity Detection) in xten products.
d the 'calling' DTMF string is sent to the calling
party. Both parameters can be used alone.
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ou can find)
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harry gaillac wrote:
> nobody has an answer here!
Actually someone asked for you config details.
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Miloš Kocbek wrote:
> Yes but i want to enable access for all users from that ip address. I
> don't want to write every user in sip.conf.
So use no secret and host=x.x.x.x where x.x.x.x is the IP address you want to
allow from.
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Elmar Haneke wrote:
> I would suggest chan-capi-cm for any configuration.
You know which quadbri cards it works with?
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Bartosz Piec wrote:
> Matt Riddell napisał(a):
>
>> Did you install spandsp?
>
>
> Yes, I have installed libtiff, spandsp, txfax and rxfax.
> The problem now is that asterisk doesn't disconnect but when I try to
> receive the fax, nothing happens. Fax (PSTN) i
shenanigans wrote:
> I was interested in getting feedback from current mail group users.
There is a limit to the number of times you can post this...
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h
disconnects. What can be a reason? I'm using 1.0.9 version.
>
Did you install spandsp?
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're guaranteed they'll use it).
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Bartosz Piec wrote:
> Matt Riddell napisał(a):
>
>> -1
>
>
> I've tried it. It just leaves the last 1 digit and drops the rest.
>
You could try ${EXTEN:-LEN(${EXTEN:1})}
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Bartosz Piec wrote:
> Hello,
>
> {$EXTEN:1} is used for dropping the first digit. But hot to get rid of
> the last digit? Is it possible?
>
-1
:)
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That way if the remote DNS goes down
you'll only lose updates.
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htt
[EMAIL PROTECTED] wrote:
> I tend to agree with you, my experience with Teliax has been decent,
> and getting better. If only I could get to them at under 20ms though,
> right now my latency is about 75ms whereas voipjet comes through at
> 19ms.
Where are you located?
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Sorry, what exactly does your spam have to do with Asterisk Users?
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Tom Vile wrote:
> Does freevoip support other codecs other than GSM?
Not at the moment. What would you like to see?
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ble and
via gotoif for the various statuses (including "beforerun" which would mean
that the AGI didn't run at all).
While this doesn't exactly answer your question, it is the best way to use
multiple statuses.
Make sense?
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Francesco Peeters wrote:
> Hi all,
>
> Since a few days my (*) no longer seems to log in to FWD through IAX2.
Use freevoip instead:
http://freevoip.gedameurope.com
(It links into FWD when FWD is up)
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x27;ll probably have to use on of the flags in the dial command that
keeps Asterisk in the loop.
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eting, please don't hesitate to let us know.
If you don't want to join up with FreeVoip you can just dial
(IAX2/freevoip.gedameurope.com/691)
We've got dev's standing by to help people!
:D Let's get this beta process done and onto 1.
Password = does not exist.
The reason it is probably working is that by using that you are specifying no
password for the account. Check your settings.
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http
Michael J. Lynch wrote:
> Matt Riddell wrote:
>
>> I just tested it and it's working fine.
>>
>> Does your Linux box have internet access?
>>
>
> Yep, but through a firewall. I figured it probably works ok and
> that I must just be doing something
I just tested it and it's working fine.
Does your Linux box have internet access?
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s a lot (about 100 MB per Month)
Have a look at the CLI command logger rotate and the asterisk -rx command
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Danny wrote:
> http://www.stellenboersen.de/stellenboersen/
Can we get this person removed from the list. This is pure spam.
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n* be good. I am part of the Vexi Project (http://www.vexi.org)
> fork
> from Ibex (nee XWT). This was a good fork. OpenPBX could be another good
> fork. Only time will tell.
Agreed. It's nice to have a measured rather than inflammatory post on the
subject (I
*PLONK*
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great loss if they subsequently leave?
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--Bandwidth
..
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more spamming
from the OpenPBX crew.
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--Bandwidt
} ${CALLERID})
> exten => ${IPCCIDN20},2,SetCallerID(${CALLERID})
> exten => s,3,Answer
> exten => s,4,Goto(main-menu,s,2)
> exten => s,5,Hangup
First there is no s,1 or s,2
so you will never get to s,3
Secondly what does the ${IPCCIDN20} variable contain?
--
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unbelievable, and the
358456347563th one crazy!
:D
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__
trixter http://www.0xdecafbad.com wrote:
> Anyone thinking about doing a VoIP business may want to get more info
> before proceeding since they may not have the millinos vonage has to
> fight this.
Unless of course they don't live in the United Sue'ers of America.
:D
--
Ch
Angus Comber wrote:
> Hello
>
> In my extensions.conf file:
>
> [frompstnisdn]
> exten => s,1,Dial(SIP/200&SIP/202,20)
> exten => s,2,Voicemail(su200)
> exten => s,3,Hangup
If you really want to use s, you will need to add an extension:
exten => 78736
he reception phone.
Asterisk will not cause it to wait two or more minutes. 3 seconds yes, 2
minutes, no...
Unless you have some funky gotoifs or loops or waits etc..
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Sergio Serrano wrote:
> Hi Srs.,
>
> Do you know if it's possible make a videocall from asterisk to UMTS
> mobile phone?. Both technologies use H.263 like videocodec.
Not yet, working on it.
--
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Bruno De Luca wrote:
> Correct: Use ${DNID} to get the number. I'm sorry.
> Bruno.
${CALLERIDNUM} is the most commonly used.
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http://www.s
should work.
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an my screen
width whereas the old versions are long, but not too wide.
Each to their own!
:D
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Michiel van Baak wrote:
> On 02:57, Thu 22 Sep 05, Matt Riddell wrote:
>>Tomasz Chmielewski wrote:
>>
>>>How can I manipulate the incoming callerID number (and add 0 before it)?
>>
>>exten => s,1,Answer()
>>exten => s,2,SetCIDNum(0${CALLERIDNUM})
&g
Adam Robins wrote:
> I have two Asterisk boxes that I thought were trunked, but based on not
> seeing the (T) in iax2 show peers, now I'm not sure.
Make sure you have some form of Zaptel timing (i.e. Digium Cards/ZTDummy)
--
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M
Tomasz Chmielewski wrote:
> How can I manipulate the incoming callerID number (and add 0 before it)?
exten => s,1,Answer()
exten => s,2,SetCIDNum(0${CIDNUM})
exten => s,3,...
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how peers in the console, it should show a (T) for trunked
connections.
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ng from many to one or one to one?
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s (ACKS maybe) are arriving late. Sufficiently so
that Asterisk is about to retransmit the packet. However, right at the last
minute it got the ACK from the last one and stopped the retransmission as it
found the ACK.
Just a guess mind you.
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Chris Miller wrote:
>
> I'm looking for advise on troubleshooting QOS problems. After much
Have a look at SineStatIAX:
http://www.sineapps.com/sinestatiax.php
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o use.
>
> Can someone point me in the right direction ?
My Asia/Pacific distributor should contact you shortly.
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k can recognize different rings.
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> Any other suggestions please??!
Send me a copy of a wave file with the recorded beeps after hangup and I will
see if the tones are somehow different.
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h
all on (03) 4555770
(Dunedin) if you are still having problems and we'll get it sorted out.
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ign (i.e. +/-)
That would give you possible values between -127 and +127.
So, a max value of 63 would either indicate a signed 7 bit variable (dunno
where the other one went) or an unsigned 6 bit variable.
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google at
http://www.google.com - it's a search engine)
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_
Sascha Ferley wrote:
> Anyone know how to reset this phone or force it otherwise to download from
> TFTP ? Anyone else experience this issue?
Is it grabbing its IP address from DHCP?
Are the other phones?
Is the tftp server on the same machine as the DHCP server?
--
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s?
Use SineStatIAX to make a test IAX call and it will measure packet drops OOO,
jitter buffer changes etc:
http://www.sineapps.com/sinestatiax.php
It requires the .net framework version 1.1 (but if I remember correctly I put
a routine in the installer to check for it and download it if mi
/1-1'
> == Starting Zap/1-1 at zap-custom,s,1 failed so falling back to exten
Could you show us your zap-custom context?
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Joseph wrote:
> For those of you who wanted to know how the AG-468 (4xFXS) unit work (or
> it doesn't work), here is my personal experience.
>From what company?
Is it PA1688 chip based?
What firmware version? 1.44040?
What protocol? IAX2/SIP/OpenH.323/MGCP etc etc
--
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cy redundant?
:)
Use different contexts if you want to split by companies, then failover to
other PCs if you have hardware/network problems.
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hat if you were using a system
like this then the variable would only contain the status from the last entry
and only then if all of the entries failed.
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Mark Phillips wrote:
> You want to speak with Matt Riddell. He's on the list
Indeed I am!
Call me 24 hours +64 3 4555770
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Adam Robins wrote:
> Should it be in half duplex or full duplex?
Full.
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nality in different way but really want
> that sort of dial plan is that possible or i am asking a dumb question
Asterisk Realtime Architecture.
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime
http://www.asteriskdocs.org/modules/news/article.php?storyid=28
have any info on this? I'd be quite interested for some projects I'm
working on too.
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d agree with you.
/me == always the diplomat.
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nect the originated call with an extension/context
so that once that number answers it is connected to say an agent or an
application. This part should be pretty self explanatory.
Make sense now? Feel free to ask if it doesn't!
:)
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__
Dennis Gilmore wrote:
>>But then, you listed your minimum price as $1. What do I get for this?
>
> :D A one minute phone conversation. :D maybe a bag of chips.
Is delivery included? If so I'll have 3. Send them to Dunedin, New Zealand.
hehe
--
Chee
trixter http://www.0xdecafbad.com wrote:
> On Mon, 2005-08-29 at 15:24 +1200, Matt Riddell wrote:
>
>>trixter http://www.0xdecafbad.com wrote:
>>
>>>controlplayback seems to fit if all you want is mp3s however ...
>>
>>Although it works with all sup
broken from.
Not the greatest idea Rob.
Why don't you just provide the instructions and link to the readytechnology
site.
You're treading on thin ice and I'd hate to see the PA1688 resources you have
hosted there disappear.
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till i need you
> people to give suggestion to improve it
>
> Primary task of this application is to get a parameter
> from extensions.conf, query sql server and play a
> files according to the result
Is this GPL?
Is there a site where people can read about it a
rations.
BTW: Rather than checking for all of the possible failures that could occur
why not just check for a success or lack thereof.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.c
00
Even at $50 it would still be 10 times more than your max!
:D
But then, you listed your minimum price as $1. What do I get for this?
:D
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Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.s
Adam Robins wrote:
> We are in the process of an Asterisk call center deployment using IAX2
> G711 ulaw softphones. Outbound sound quality is terrible.
Check if the network card is in half duplex mode.
--
Cheers,
Matt Riddell
___
trixter http://www.0xdecafbad.com wrote:
> controlplayback seems to fit if all you want is mp3s however ...
Although it works with all supported formats.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
h
en => 1234,1,ControlPlayback(file|4000|*|#|1|0|5)
--
Cheers,
Matt Riddell
___
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http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
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Bill McCready (PCPhoneline.com) wrote:
> If you have any questions, please private e-mail me. Thanks !!!
>
> Best regards,
>
>
> Bill McCready
> PCPhoneline.com
This is spam. Post it to the Asterisk-Biz list. This list is for
"Non-Commercial Discussion"
e to send/receive text messages
without having to actually be in a call.
--
Cheers,
Matt Riddell
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