Christian,
Thanks for your response.
In my case, I was asked to do it through SIP phone 3 way call functionality
and not the Asterisk Meetme application.
I wanted to know if any one had done something similar in past or not.
I am short of PRI in my test environment and hence I can't test it
On Monday 20 Dec 2010, Olivier wrote:
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
(written by someone else before me) which sets up calls by creating
files of
the general form
Channel: SIP/$INSIDE_NUMBER
Hi
Looking at the source of app_voicemail.c there are many statements like:
ast_debug(1, %s doesn't exist, doing what we can\n,
prefile);
Where do I have to enably this to be showed in the console or logged to a file
by logger. core set debug does not seem to help
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death), or
the kernel runs out of PID
On Tue, Dec 21, 2010 at 11:47:02AM +0100, Benoit Panizzon wrote:
[snip]
Well, my actual problem is, that if a customer has recorded his own greeting,
he usualy tells the caller to record his message after the tone, so
app_voicemail should not play the intro.
spool/mailbox/unavail.gsm
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
;;exten=_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=_8[1-9],n,Hangup
And that works.
Also changed the hints:
;;exten = 81,hint,MeetMe:81
exten =
Hello
I've done some googling, but still puzzled at my working
configuration.
Apparently, a user can only receive calls through Asterisk, a peer
can only make calls, and a friend can do both.
If that's correct, I don't understand why my VOSP requires the
following settings in sip.conf
On Mon, 20 Dec 2010 12:39:44 -0600, Kevin P. Fleming
kpflem...@digium.com wrote:
You've missed a very important point here: you are using a *SIP*
endpoint to call a *SIP* URI. The endpoint can do that directly, and
doesn't need any help from Asterisk to do it. If you wanted to be able
to
On 12/20/2010 07:08 PM, Dovey Forman wrote:
Thanks Kevin.
Did it work with Asterisk 1.2 because it ignored it?
I don't know specifically that Asterisk 1.2 ignored Required headers,
but it's certainly possible.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan
On Tuesday 21 Dec 2010, Gilles wrote:
But I could use a good article/book to better understand my options,
how Asterisk is different from the alternatives (Freeswitch, openSIPS,
etc.)
www.amazon.com/s/ref=nb_sb_noss?url=search-alias%3Dstripbooksfield-keyword
s=voip
The same way Ubuntu,
On 21 Dec 2010, at 14:20, A J Stiles wrote:
Well, every Free and Open Source telephony system is using Asterisk (and
Linux) under the bonnet. The differences are in the user configuration
tools.
Uh, no?
S
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On Tue, 21 Dec 2010 14:20:55 +, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
The same way Ubuntu, Slackware, CentOS c. differ from each other. They are
all using the Linux kernel and the X Window System under the bonnet. Well,
every Free and Open Source telephony system is using
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
[conferences]
exten=_8[1-9],1,Answer()
Am 20.12.2010 21:39, schrieb Ernie Dunbar:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either
the Asterisk server is restarted (and the zombies die a natural death),
or
the kernel runs out of
Your server is being brute-forced. Read this article (http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk) and set up fail2ban on your machine right now.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS -
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until either
the Asterisk server is restarted (and the zombies die a natural death),
Actually, no. This is part of a migration, and those are mostly customers'
secondary lines (which for the most part, aren't even active). We get a
lot of these bad logins because the retry times on the ATAs are quite
short.
Asterisk really *shouldn't* leave zombies around for every bad login, but
Hi Everyone,
I understand that there are a few warnings about using cp to move .call
files to /var/spool/asterisk/outgoing as they might acted upon before copy
is done. So, using PHP, What is the equivalent of mv command? Would it be
rename() in php or is there a better method?
Thanks,
--
PERL has a move() command; I wouldn't expect less out of PHP.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Tuesday, December 21, 2010 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I think rename() is what you're looking for
http://php.net/manual/en/function.rename.php
On Tue, Dec 21, 2010 at 2:23 PM, Danny Nicholas da...@debsinc.com wrote:
PERL has a move() command; I wouldn’t expect less out of PHP.
From:
I have been using:
exec ('mv *.call /var/spool/asterisk/outgoing')
and for a long time it has been working just fine for me on more than one
websites. Just make sure the folder where you create the call files has
correct permissions and ownerships so that the file is successfully moved by
the
2010/12/21 Paul Belanger pabelan...@digium.com:
On 10-12-20 05:51 PM, Jarek Jarzebowski wrote:
OK, so I have attached debug log.
I am using:
*CLI core show version
Asterisk 1.8.1.1 built by root @ asterisk on a i686 running Linux on
2010-12-17 23:03:58 UTC
Definitely a bug, ran into the
On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either the Asterisk server is restarted (and the zombies die a natural
death), or the kernel runs
On Tuesday 21 December 2010 04:49:42 A J Stiles wrote:
On Monday 20 Dec 2010, Olivier wrote:
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial
application (written by someone else before me) which sets up
calls by
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
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On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to include:
_NXXNXX
_NXX
_011.
_911
into my current plan:
Sorry, here's the rest.
exten = _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten =
On Mon, Dec 20, 2010 at 5:39 PM, Ernie Dunbar maill...@lightspeed.ca
wrote:
We have an issue with our Asterisk install where Asterisk produces many
Zombie processes (on the order of several hundred per minute) until
either
the Asterisk server is restarted (and the zombies die a natural
In a 1.4.24 system, out of several lines, one of ours gets 1 or more random
calls a day with Asterisk as the caller ID.
I have just seen this described in the last couple of weeks, but at the time
it wasn't happening to us, and I the explanation didn't stick with me.
Can anyone give me a pointer
Cary Fitch wrote:
In a 1.4.24 system, out of several lines, one of ours gets 1 or more random
calls a day with Asterisk as the caller ID.
In my experience, it happens when the caller is blocking their CID. I
have programming in place that assigns the named restricted and the
phone number
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm trying to migrate from MeetMe to ConfBridge:
On 12/21/2010 10:03 PM, sean darcy wrote:
On 12/21/2010 12:13 PM, Jeremy Betts wrote:
What version are you running?
I believe device state tracking for ConfBridge was recently added.
On Tue, Dec 21, 2010 at 3:39 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
I'm
On Tue, 21 Dec 2010, Bruce B wrote:
So, using PHP, What is the equivalent of mv command? Would it be
rename() in php or is there a better method?
Not really an Asterisk question...
On Tue, 21 Dec 2010, MrHanMan wrote:
I think rename() is what you're looking for
+1
On Tue, 21 Dec 2010,
Anyone??
Thanks.
On Mon, Dec 20, 2010 at 10:42 AM, VoIP Question voip.quest...@gmail.comwrote:
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
2010/12/18 Matt Riddell li...@venturevoip.com
On 17/12/10 5:56 PM, Olivier wrote:
Hi,
Did you use libpri 1.4.11.5 or 1.4.12-beta ?
Recently l tried 1.4.11.5 on a live system and it failed (Asterisk
1.6.1.18 and dahdi trunk, Junghanns QuadBRI, PtmP lines).
Going back to 1.4.11.2 solved
Hi Stephen,
_NXXNXX
_NXX
_011.
_911
Of course it can, but it depends on what you want to do when those
numbers are called...
I didn't know about the setvar in the sip.conf and actually I think it
is a much cleaner solution. Since you are already using it I would
suggest to not only
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero on
'SIP/Proxy-0031'
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