brings up a good point. We've just reverted a couple of
installs from 1.6.2 because of deadlocks. What version should we be
going to?
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is login and run
dahdi_cfg
/etc/init.d/asterisk restart
but that's a pain to have to do after every reboot. I've never had this
Don't know why it's happening, but add those lines to /etc/rc.local as a
quick hack in the interim until you find out what's causing it.
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and not on demand?
You could compile Asterisk with embedded modules?
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!
It must not be running that line - have you done a dialplan reload?
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On 16/03/11 5:43 PM, Nikhil wrote:
ok..that means I have to modify chan_sip . I wondering why this is not
available in asterisk.
Because you haven't completed the patch yet! :P
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(${somecommand})
and use the manager to set the somecommand variable on a call you send
to the dangerous context.
Up to you.
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On 11/03/11 7:52 AM, Nick Ustinov wrote:
These are the same for sip users and trunks
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
Who is asking to transmit frame type slin ?
Maybe transcodeviaslin or something with a Local channel?
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but no go.
The cards both work with mISDN and chan_lcr, but we get reasonably
frequent crashes.
Does anyone have BRI working at all with the latest Asterisk, DAHDI, LibPRI?
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On 4/03/11 12:15 AM, Dan Journo wrote:
Hi,
Does anyone have a good VoIP Bandwidth Calculator?
http://www.asteriskguru.com/tools/bandwidth_calculator.php
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with the
lower qualify time?
Traditionally you'd use a value you consider to be good enough for calls
and set qualify to that. I.E. if you think 30ms is ok then set
qualify=30 and then just route via the first then the second depending
on status.
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,
Matt Riddell
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.
I'm going to try again this weekend - with a different b410P card - even
though it works with mISDN :)
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On 11/02/11 6:54 PM, William Stillwell wrote:
I was getting unable to make channel..
We couldn't get it to work properly until we upgraded to Asterisk 1.8 at
which stage it magically started working (with the same configs etc).
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in the website, pretty much same location, phones look the
same etc etc.
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[20419]: chan_dahdi.c:12946 pri_dchannel: PRI
got event: HDLC Abort (6) on Primary D-channel of span 1
If we change the hardhdlc to dchannel instead the message goes away, but
obviously it doesn't work :)
So, anyone have any ideas?
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up for the next answer etc - crazy.
Much easier when replies are inline with the questions.
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were then generated.
Heh, latest everything - so LibPRI trunk.
I did try going backwards in terms of DAHDI, but not LibPRI - will try
that on Monday.
By the way, Kevin/Russell etc, any chance we could get a test added to
bamboo for physical connectivity?
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==
Anyone have any ideas? Obviously it doesn't receive or make any calls -
pri is in provisioned, down, active.
I had to reinstall the old box (mISDN etc) temporarily until we can get
this sorted.
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==
Anyone have any ideas? Obviously it doesn't receive or make any calls -
pri is in provisioned, down, active.
I had to reinstall the old box (mISDN etc) temporarily until we can get
this sorted.
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) it is a dead project.
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in
the console?
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see much better success
from it, because people would redirect there.
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or suggestions, feel free to mail me on them.
Oh, and we've moved the Daily Asterisk News web server to Dallas, TX, so
it should be a bit quicker for those of you in the states - well,
anywhere except New Zealand really :)
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and email, then posting back here with your
results :)
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).
This has happened at least 20 times in the past two days. At first the
supervisor thought that the same call was ringing on three different agents at
once but the logs say that the first two do not answer and the third does.
What strategy are you using for the Queue?
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. But it
happens
very regularly, too often i would say.
Try it with Zoiper and see how you go. I've not seen the same thing happen.
It may also be that you are using qualify and that the peer is too far away.
What do you get when you type iax2 show peers?
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On 3/09/10 8:32 AM, Arnaldo Giacomitti Junior wrote:
There´s a way to get the channel signalling in dialplan?
I have changed the code in channels/chan_dahdi.c and includes:
Upload it as a patch to the issue tracker:
http://issues.asterisk.org
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,PrivacyManager(3,10)
exten = jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
exten = jw,n,Verbose(-- CID is${CALLERID(num)})
exten = jw,n,Dial(SIP/1000,60,w)
Maybe you could do:
Set(CDR(userfield)=${CALLERID(num)})
Before dialing SIP/1000
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something happening when the call gets cut off.
Is there any DTMF being transmitted, why was the call disconnected etc.
Or just take a snippet and put it up on pastebin/post here
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of the call, I lose audio from the
asterisk box to my soft phone, but not the other way around. This looks
like one commit, but obviously I would like to know what's going on
here?
What's in the commit?
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were that incoming calls
would not hear any ringing tones and the call would be ended by Teliax
after 21 seconds.
You could just answer the call before dialling your internal extensions.
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in there - just make sure the
username and password are correct in the user's device.
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=no ; Disable IAX native transfer
;transfer=mediaonly ; When doing IAX native transfers, transfer
; only media stream
Well, it depends on what version. The above is from a 1.4 system -
earlier systems had notransfer=yes, but not the mediaonly option.
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not happy)
I really thought that the canary should have sounded if Asterisk got in
a loop - or maybe that only happens with high priority?
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On 20/08/10 1:52 AM, Tino wrote:
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Record?
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billsecduration;
+--+
| COUNT(*) |
+--+
|0 |
+--+
1 row in set (0.31 sec)
mysql SELECT COUNT(*) FROM cdr;
+--+
| COUNT(*) |
+--+
| 190052 |
+--+
1 row in set (0.00 sec)
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, then there is no problem.
Has anyone seen anything like this ?
Heh, seems impossible!
Um, maybe the voicemail beep is the same tone as a * and * is used to
disconnect a call or something?
Try doing a SIP debug and see what turns up. Also make sure it's 100%
repeatable :D
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On 25/08/10 7:14 PM, Tino wrote:
Yes, we need to record the message
:D So use the Record() application :D
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. If it's something else, like
memory swapping, there's nothing the canary can do to fix that.
Aha, explains why I've never seen the canary die :D
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is) and
priorities which increase. It makes some things a bit harder, but you
can always use labels and the read application.
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,
Feedburner, Amazon EC3 etc etc.
Sure you have to decide who you want to trust (personally I trust the
humbuglabs guys) and what their level of protection is (are they looking
after their own security), but it seems to be the way things are going
at the mo.
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that as ALAW.
That would be signed linear (SLIN).
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does show in sip show peers and the softphone (twinkle) shows a
Registration Fails with a 603 denied.
So I'd say it's working
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I should probably try the 1.6 series
Are you using deny before permit?
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, but SecureCRT with
tabbed SSH windows and buttons which can be set up for things like nano
/etc/asterisk/extensions.conf make life pretty simple.
On Mac I now use iTerm (similar thing).
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would appreciate any feedback on writing this.
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+Wake-Up+Call+PHP
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way to press buttons.
Why not just use followme for everything but the car, and if that fails,
send the call to the car normally?
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)
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chan_ooh323.c:1938: error: dereferencing pointer to incomplete type
chan_ooh323.c:1940: error: dereferencing pointer to incomplete type
chan_ooh323.c:1943: error: dereferencing pointer to incomplete type
Do you need OpenH.323?
If not, run
make menuconfig
and disable it
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to my cell number.
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and then the patch can be added to the base
Asterisk install (assuming it meets coding guidelines etc).
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the first word - some VoIP providers take a
while to pass audio - might be that there is a delay in your dialplan or
that the first words of audio are simply not transmitted.
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might be too high - do a few tests to your
own phone and make sure it recognizes the individual words.
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Just try running:
asterisk -vcd
And you'll see the error.
Alternatively you can edit /etc/asterisk/logger.conf to allow you to
have a full log.
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you want to use (bear in mind that
they might still use an upstream provider who uses G.729 etc).
Easiest option is to just choose aLaw or uLaw based on your country.
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the
download link.
When done from an iPhone it brings up the app with a link to download,
on my Mac it opens iTunes to the application page.
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Hi all,
I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.
Hope you enjoy it :D
http://www.venturevoip.com/news.php?rssid=2371
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an option (r IIRC) to provide ringing
instead of music on hold.
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On 19/03/10 1:19 PM, Adrian Marsh wrote:
Hello,
I’m looking for some advice on securing Asterisk.
Have a look at fail2ban:
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
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the wrong way.
If you originated to a SIP device and sent the other end to the
application PlayDTMF, then it would be sent to the SIP device (if that's
what you want).
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Also, why are you saying your name is Philip?
On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote:
My dear friend Matt Riddell insists that the Manager only can dial 5
calls per seconds, which I find ridiculous. Is there a way to prove
him wrong and have him lift the limit that has
The responses from the Asterisk manager on your machine start
providing responses of no account code when calls are initiated at a
higher rate.
On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote:
My dear friend Matt Riddell insists that the Manager only can dial 5
calls per
Asterisk can fill out the details.
Apologies for top post, laptop is running a defrag.
On 24/02/2010, at 9:32 AM, Tommy Botten Jensen
tommy.jen...@freecode.no wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Olle E. Johansson skrev:
23 feb 2010 kl. 20.18 skrev Matt Riddell
Jensen tommy.jen...@freecode.no
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Matt Riddell skrev:
Yeah, the problem's not the origination.
The problem is that calls originated asyn with accountcodes show up
in
show channels concise without details.
Pretty simple to test
, not in the way you
expect. Likely if you did a query for exten = 100, the n extensions
would be returned in a random order.
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if this means no solution for me.
Then I know it's not doable.
:)
Maybe you should read the messages from the list then :)
You've already been replied to.
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?rssid=2353
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there may be a coding mistake). Thus what do you prefer to do? How
can i examine the core dump file?
http://www.voip-info.org/wiki/view/Asterisk+debugging#CoreSoftwareDebugging
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):
prepend domain-name-servers 127.0.0.1;
Otherwise, your entry in resolv.conf will be overwritten on each DHCP
lease renewal.
Yeah, although if you're using DHCP, then dnsmasq is possibly a better
option.
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Then it's done.
Dnsmasq is probably overkill for this type of thing, though some people
in the office prefer it to bind.
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}+300,int)})
exten = 3,n,WaitExten(3)
exten = 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does 300 and 4 does 500.
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he/she'd rather just bitch.
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it
done :-(
Not that I'm aware of - best place to ask would be the IAXClient mailing
list, but I'm pretty sure I'd remember if someone had written one.
Probably the closest would be Tim Panton's work - maybe hunt him down :D
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What user are you running Asterisk as?
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On 16/01/10 12:56 AM, nak...@02.246.ne.jp wrote:
Hi, I have a question about jitterbuffer and PLC.
Do you get the same results if you use:
iax2 test losspct x
Where x is the loss percent you'd like to test?
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On 29/12/09 10:22 AM, Leif Neland wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Use some of the Chinese PA1688 or AR1688 phones - support auto answer,
IAX/SIP etc.
Prices around $45
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.
If the calls between VoIP-VoIP are too loud and the calls to PSTN are
too quiet, then likely the provider needs to check their gain settings.
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it :)
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? I.E. Gnome/KDE/XFCE etc
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clarification - I wouldn't say it's defences.
By default these calls are sent to the default context (which should not
have the capability to make calls other than test the system).
So, yes you are allowing unauthenticated calls, but to the echo test etc.
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- so they're safe.
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it up for discussion only to hear it was
the wrong way to do it.
At least the patch is small :D
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On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
Maybe the best way would be to make it that the default context only
provides the info from the examples unless you provide an option:
read_security_document=yes
Asterisk used
exactly the same problem when
using an old analog phone with a Linksys SPA-3000 instead.
Has anyone encountered this problem before? If so, what caused it and
what solved it?
Are you binding to an address that the box doesn't own?
Check the top of sip.conf.
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/cdr_mysql.conf
Also, the status check is cdr mysql status
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has idea about Grandstream, and if they have a lot of problems and
such noise in handset? Or my luck was bad that this phone is defected?
I wouldn't recommend the BudgetTone - it's been a while since I used it,
but there are better phones around (even from Grandstream).
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On 10/11/09 1:02 PM, Conklin, Tom wrote:
Have you taken a look at the following?
http://www.astassistant.com/
Also:
http://www.asternic.org
and the newer version:
http://www.fop2.com
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for send_sms is:
1. Read AGI variables
2. Get destination variable
3. Include clickatel API file
4. call send_sms function
We also provide an API from our telephone exchanges, but to be fair
you're likely better off just using clickatel yourself :D
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. Set up sip.conf/iax.conf based on what type of softphone
3. Download a softphone - I've listed a few here:
http://www.venturevoip.com/news.php?rssid=2188
4. Make calls :D
The most important step is number 1 - once you get the hang of Asterisk
the rest will be easy :D
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to. That way if someone is getting overloaded with
support requests you can move jobs to another staff member.
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wanting is more on the jabber client side -
you're wanting one that can receive messages and display them as pure HTML.
There may be one - I don't think Adium (the client I use) does it, but
if you had a look at a few different clients, maybe one will.
--
Cheers,
Matt Riddell
Director
(much
the same way that the followme app does).
In fact it sounds like what he's actually wanting is the followme app:
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
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Cheers,
Matt Riddell
Director
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is away from his desk
* Jane dials *79 (also in group 3) and picks up the call
If Fred (in group 5) were to dial *79 he would not pick up the call.
Names have been changed to protect the innocent :D
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Cheers,
Matt Riddell
Director
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http
the commented line // with the line in question)
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Cheers,
Matt Riddell
Director
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On 23/10/09 6:11 AM, jonas kellens wrote:
On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote:
It's really simple you just read from standard input and write to
standard output.
If you tell us a programming language you'd like to use (i.e.
php/c/perl/bash etc) we can give you a link
a shot - all boxes using NTP?
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Cheers,
Matt Riddell
Director
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really? Where would I find that?
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Cheers,
Matt Riddell
Director
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brings me to another question - what does Digium recommend people
use on a 1.4 system with their b410p card these days?
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Matt Riddell
Director
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, and what's going wrong?
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Cheers,
Matt Riddell
Director
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, unless you specify Asynchronous mode using Async: true”. Guess
I’ll never be as smart as you, Matt.
:D
I should hope not!!
If everyone was as smart as me, how would I take over the world?
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Matt Riddell
Director
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