Is anybody else using the following combination:
* a TE410P card (wct4xxp driver)
* a BT ISDN connection
* DAHDI 2.3.0.1
* Asterisk 1.6.2.9
I'm trying to configure a new box to replace a legacy system (same hardware;
some old version of Asterisk with Zaptel; works lovely but hopelessly
On Tuesday 22 Jun 2010, Tzafrir Cohen wrote:
The old ones should work just as well. Apart from 'echocanceller' lines
in system.conf. Those may prevent you from having a working echo
canceller, but nothing worse.
What do you have in those files?
What's the output of lsdahdi ?
Files
On Tuesday 22 Jun 2010, Tzafrir Cohen wrote:
On Tue, Jun 22, 2010 at 12:26:28PM +0100, A J Stiles wrote:
Is anybody else using the following combination:
* a TE410P card (wct4xxp driver)
* a BT ISDN connection
* DAHDI 2.3.0.1
* Asterisk 1.6.2.9
I'm trying to configure a new box
On Tuesday 22 Jun 2010, Scott Stingel wrote:
Hi-
I've been going through the same upgrade process recently, and had the
same error (shown in your other message). I had forgotten that the
equipment I was plugged in to was CPE, so I had to change my new setting
for that span to NET rather than
Right. I think I might be getting somewhere.
First I commented out all the lines relating to spans 2, 3 and 4 in
my /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf, and set up a
very minimal dialplan in /etc/asterisk/extensions.conf (just 2 extensions).
Then I connected up just span
On Tuesday 22 Jun 2010, Mike wrote:
Hi,
I have the following happen to me after the restart of one of my servers:
out of my 3 PRIs (all configured with the same technical settings), the
last one isn't coming back. It's underutilized (chances it didn't get a
call since my reboot), if it
On Tuesday 29 Jun 2010, Tarek Sawah wrote:
. is it possible to
force the agents (users) to use a certain UserAgent which is the one
built-in our system? this way will prevent the agents we are restricting
them to only be able to dial through the software which is already
restricted to
On Friday 02 Jul 2010, Ira wrote:
At 11:14 PM 7/1/2010, you wrote:
Same activity from these IPs:
174.129.137.135
Given that my Asterisk box is used for nothing but Asterisk and I
know the small number of IPs that need to have access is there an
easy way to use iptables to block everything
On Friday 02 Jul 2010, Tim Nelson wrote:
- A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Friday 02 Jul 2010, Ira wrote:
At 11:14 PM 7/1/2010, you wrote:
Same activity from these IPs:
174.129.137.135
Given that my Asterisk box is used for nothing but Asterisk and I
On Tuesday 06 Jul 2010, ABBAS SHAKEEL wrote:
Hello Community,
. I am facing an issue of security i.e. We deploy
servers to client end. Now i dont want the client to see my configuration
files (Of course copy and distribute or replicate the logic with out
permission). [ 1 paragraph
On Friday 09 Jul 2010, Christian wrote:
Hi all,
Yes, this is not the right list for such a question and I am using Asterisk
myself its for a friend who isn't used to Linux. You can write me off list
if you want. He is looking for a Windows based PBX with same functionality
as Asterisk. Any
On Tuesday 13 Jul 2010, Randy R wrote:
I was thinking of closing port 25 and using an alternate port (587?)
setup if the spam service is able to connect to an alternate port.
That way, the users can also change their configs to 587 and most
spammers will be trying 25 which is closed.
Can't
On Friday 16 Jul 2010, Neeraj Chand wrote:
Hi All,
After getting licences for Skype for asterisk a while ago I finally
got
around to setting up a server with two channels and setting up a bcp
on
the skype end.
The idea behind this is the following:
Users can dial into the PBX,
On Monday 26 Jul 2010, Andraž wrote:
Hi,
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed freetds-common,freetds-dev, libct4, libsybdb5,
freetds-bin, but, when I run configure and then make menuconfig in section
Call Detail Recording - cdr_tds it's
On Friday 30 Jul 2010, Andraž wrote:
From source also doesn't work. :(
If you ran ldconfig to force update of library configuration after you
installed the freetds you compiled, and re-ran ./configure in the asterisk
build directory, and it still doesn't want to let you use freeTDS, then
On Monday 02 Aug 2010, Janu Mukherjee wrote:
Hi all,
I have the following problem. I want to
Call -- Asterisk AGI Answer -- Create File - Copy File Asterisk
-- Play File -- Finish Call
For now we are using sshfs to map the directories. I now want to achieve
this using samba server. I am
On Tuesday 10 Aug 2010, Gilles wrote:
Hello
I just read an article on the tiny Ben NanoNote:
http://en.qi-hardware.com/wiki/Ben_NanoNote
As CPU, it uses a JZ4720 366 MHz MIPS compatible processor from
Ingenic Semiconductor Co, and it runs Linux.
Does someone know if Asterisk has been
Hi
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding the
value in the `duration` field.
I didn't think this was supposed to happen? Our old installation (some
ancient version, sorry not available)
On Tuesday 17 Aug 2010, Jason Morgan wrote:
Hi,
I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support)
at the same
time as moving from Ubuntu hardy to
I have a single TDM400P rev I with two fxo and two fxs modules, these were
working perfectly for years
on Asterisk 1.4
On Wednesday 11 Aug 2010, Tilghman Lesher wrote:
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding
the value in the `duration` field.
I'd
On Tuesday 24 Aug 2010, Zeeshan Zakaria wrote:
I am planning a migration to virtual machines, and was considering with it
to move from 1.4 to one of the later versions. My and my clients' 1.4
setups have been rock solid and I don't want to put myself into any
unnecessary trouble. Those of you
On Wednesday 25 Aug 2010, Doug Dawson wrote:
Todd
To interface directly with the telco pots lines You should be using FXS
modules with FXS signaling.
No. FXO is what you need to connect to a phone line. FXS is to connect to
and ring an analogue telephone. (S = Signalling; i.e. it can
On Thursday 02 Sep 2010, Ken D'Ambrosio wrote:
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go
On Thursday 16 Sep 2010, Nikhil wrote:
Hi
I got the bellow error when I try to configure asterisk code.
$./configure --with-ssl=/usr/local/ssl
...
...
...
checking for mandatory modules: OPENSSL... fail
configure: ***
configure: *** The OPENSSL installation appears to be missing
On Thursday 16 Sep 2010, Tim Nelson wrote:
I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX
2.6.0. There are a large number of inbound routes configured for the
various DID's coming in via PRI, SIP, etc. If a user calls outbound to one
of these numbers, it goes out to
On Wednesday 29 Sep 2010, Lee, John (Sydney) wrote:
Do you mean that if I could define 30 channels in span 1 for example, then
span 1 is set to E1?
If not, then it is T1.
Yes! That's how it works.
Civilised countries manage to squeeze 30 B-channels and two D-channels onto an
ISDN line by
On Thursday 30 Sep 2010, Danny Dias wrote:
Hello,
I'm getting a KErnel Pannic every time i restart the server, what could be
happening?
I just make: shutdown -r now and the server gets Kernel Panic. I'have to
go on site and press the power button
Here you have my sotware versions:
On Monday 04 Oct 2010, Flavio Miranda wrote:
Hi all,
Every time I reload my asterisk it fall down and the following message
appear on log:
parse error: No category context for line 7 of
/etc/asterisk/chan_dahdi.conf
If I comment that line, it change to other line.
There are some thing
On Friday 15 Oct 2010, Zarko Zivanovic wrote:
Hello,
We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
natted network.
We have the issue with calls to these SIP phones - no audio.
It is probably the problem with port forwarding on router - but I am not
sure how can
On Wednesday 01 Dec 2010, RR wrote:
Zaptel package isn't installing though ...crashes midway complaining that:
*Operating environment requirement not met.
This package requires Solaris 7 or better.
checkinstall script suspends*
huh? I'm running 5.11, which according to some rigorous
Does anyone know of a smartphone available in the UK, which is capable of
running Asterisk and has Zaptel / DAHDI drivers available for its own
telephony engine?
--
AJS
--
_
-- Bandwidth and Colocation Provided by
On Friday 17 Dec 2010, Khaled W. Chehab wrote:
HI,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
# /etc/init.d/ircd stop
# chmod -x /etc/init.d/ircd
Should do the business :)
--
AJS
--
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
Context: $CONTEXT
Extension: $OUTSIDE_NUMBER
Priority: 1
CallerId: $INSIDE_NUMBER
in
On Monday 20 Dec 2010, Olivier wrote:
2010/12/20 A J Stiles asterisk_l...@earthshod.co.uk
Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
(written by someone else before me) which sets up calls by creating
files of
the general form
Channel: SIP/$INSIDE_NUMBER
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 Tuesday 04 Jan 2011, Gilles wrote:
Thanks Sebastian for the tip. The goal is to 1) have clients call the
usual landline number instead of asking them to try a cellphone in
case no one's home, 2) get Asterisk to handle the call, 3) have the
cellphone ring with the CID of the original caller
On Tuesday 18 Jan 2011, salaheddine elharit wrote:
yes i want to know how can i do in order to read this files using apche
Either make a symbolic link to the location of the files from somewhere Apache
knows about, using something like
# ln -s /path/to/files /path/to/webroot/mp3files/
and set
On Tuesday 18 Jan 2011, Don Kelly wrote:
PLONK is retro--like bottom-posting :)
--Don
Retro? For those of us who actually know what PLONK means, it's hilarious.
The fact that some people *don't* know what it means only makes it doubly so.
Now, here is a link that those of us who remember a
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then 'asterisk -vr' to attach to the
asterisk console there are no
On Saturday 22 Jan 2011, Tim Panton wrote:
I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card.
Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable?
If so, any clues where I might buy one in the UK? The Digium card sellers
don't seem to stock such
On Monday 24 Jan 2011, Tilghman Lesher wrote:
On Monday 24 January 2011 03:46:18 A J Stiles wrote:
white/blue blue white/green orange white/orange green white/brown brown
This is incorrect. The pairs should be:
blue white/blue white/green white/orange orange green white/brown brown
Wire
On Monday 24 Jan 2011, Jonas Kellens wrote:
I keep on getting the error :
ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
(using password: YES)
[stuff deleted]
Try hostname=localhost aot hostname=127.0.0.1,
On Wednesday 26 Jan 2011, Gilles wrote:
I'd like to display CID information on users' monitor running
Windows.
I know I can run a script through the dialplan to send a datagram that
is picked up Impulse Technology's free NetCID (www.imptec.com), but
I'd rather use an open-source
On Thursday 27 Jan 2011, Gilles wrote:
I had another idea: It'd be cool if the application could either just
display CID information, or also search Outlook for a matching Contact
and open the relevant page so that the user can review/add information
for that person. Poor man's CRM :-)
.
On Friday 04 Feb 2011, Timothy Smith wrote:
Hi Users,
I have a problem with some of my mp3 files. they crash the system
(Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to
play them.
Some distros used to use mpg321 instead of mpg123 (early versions of which
used to suffer
(Putting everything back into the right order, and stripping out unnecessary
bits, for the sake of anybody searching the archives in future.)
On Friday 04 Feb 2011, Timothy Smith wrote:
On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
Try running
$ mpg123
On Saturday 05 Feb 2011, Timothy Smith wrote:
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
Can you listen to an mp3 file through the Asterisk server's own sound
card (if it has one; if not, use the -w option to write to a .wav file,
and test
This might be a stupid question, but:
If I install a new Linux kernel on a machine running Asterisk, do I have to
recompile DAHDI?
If yes, what do I have to do to get it to build just the kernel modules?
(We use Debian here. Squeeze has just gone stable, and it requires a new
kernel.
On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Bizarre bug?
I'm guessting, this is a brand new machine on its
On Thursday 24 Feb 2011, Edwin Quijada wrote:
Hello!
I bought a virtual IP line to my ISP to use with my asterisk but when I try
to connect it to my ISP tells me I can not use and I can only use with a
softphone that gives me, xlite ready configured.
Nothing says Climb me like a fence ;)
I
On Monday 28 Feb 2011, Steven Howes wrote:
'asterisk security' is a misleading subject line. Guessing someone just
scanned some IP addresses and made calls. You need what's called a
'firewall'.
Well, assuming you're on Linux then you've already *got* a firewall. Just add
some iptables rules
On Thursday 03 Mar 2011, Piotr Górski wrote:
As free I mean no subscription. I can write AGI that will query
numberingplans.com - that's not a problem... but I can query site only 20
times a day without a subscription... So it's not free.
Well, free is as free does :)
For the time being,
On Thursday 24 Mar 2011, Peter den Hartog wrote:
I would like to use the from caller id, to allow calls yes or no.
101, and 111 should be allowed to use the Trunk, the rest of the phones are
not.
Is this even possible?
So if the from caller id is 101 or 111, then allow the call, otherwise
On Thursday 24 Mar 2011, tahar .H wrote:
so plz is there any one who can give me a puch to learn this extraordinary
Asterisk plz(video things will be better :))
Start with two cocoa tins and several metres of string, and work your way up
from there .
--
AJS
--
On Monday 04 Apr 2011, Asterisk User wrote:
Hello all, I am trying to figure out the logic in on prefix matching for
Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT
calls to 011870, 01137455 and so on.
Asterisk's default behaviour is always to try the hardest-to-match
On Friday 08 Apr 2011, vip killa wrote:
Is there a way for asterisk's voicemail to send an email (including
voicemail attachment) to multiple email addresses?
It's probably easiest to set up a user on your mail server to receive the
voicemail messages that are meant for multiple recipients,
Hi. I just want to make sure I understand this before doing something that
might break things spectacularly for our users and customers :)
We are using Asterisk 1.6.2.9 and my programming language of choice is Perl.
I want, when a call comes in on someone's DDI number (which the person who
On Wednesday 13 Apr 2011, Thorsten Göllner wrote:
It should work - I think. BUT I am not really sure what will happen, if
the child process exits. The child works with a copy of all asterisk
ressources given to it, when forking. So when the child dies, perhaps
asterisk will do a hangup or
On Monday 18 Apr 2011, Niccolò Belli wrote:
As soon as the Internet connection goes down, phones
stop working. I want to be able to use pstn, isdn and the gsm gateway
even if the Internet connection goes down, how can I achieve it?
You most probably are using a nameserver somewhere else on the
On Monday 18 Apr 2011, bilal ghayyad wrote:
Hi All;
I am using Asterisk for Call Center (so agents login, logout, ready, not
ready, ... etc). To be able to have a good call center reporting, on what I
have to depend? On the CDR of Asterisk or there is another way?
Is there a good open
On Tuesday 19 Apr 2011, Niccolò Belli wrote:
A caching nameserver is not a viable solution because I want it working
even after a month without internet access.
Then just make your local nameserver authoritative for the domain in question.
You can always firewall off port 53, if the
On Thursday 21 Apr 2011, Khaled W. Chehab wrote:
Dears,
I configured an account on my asterisk pbx to record the outgoing calls.
When the asterisk pbx user make a call and send a fax the call recorded to
wave file format.
I searched the internet and found a software that can play the
On Tuesday 26 Apr 2011, bilal ghayyad wrote:
Hi All;
I am using Asterisk 1.8, how I can protect my self from hackers in case
they was able to see my sip.conf file? I need the password to be encrypted,
how?
Short answer: You can't. Asterisk itself needs to be able to read the stored
On Wednesday 04 May 2011, vip killa wrote:
Honestly Digium's Asterisk is not a quality project. Though it has lead the
way in innovative open-source VoIP, it's a flawed and chaotic project.
Hence, I refuse to pay Digium.
Don't worry. You can always get your money refunded if it breaks -- and
On Thursday 05 May 2011, bilal ghayyad wrote:
Hi All;
When the endpoint register on Asterisk or initiate a call, so they exchange
the sip username and password. What is the possibility that this will be
capture by the hacker and how to avoid this problem?
If the two devices are connected by
On Monday 09 May 2011, mahesh katta wrote:
Hi,
THIS IS IN DUBAI.
I am having PRI line with 100 DID's (00-99) and when we call to any
landline or mobile number then it shows us our board number or pilot number
(i.e 4663000 means 00)..
In the context through which outgoing calls are placed,
On Monday 09 May 2011, Cassius Smith wrote:
On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote:
Sebastian Arcus wrote:
Cisco phones (at least the 7940) are supposed to be run with a tftp
server available at all time
That is my experience. But, if you're running tftp under Linux, then
On Tuesday 10 May 2011, mahesh katta wrote:
sir,
Below configuration i wase made in server . but this is not working.
exten = _90X,1,NoOp(${CALLERID(num)})
exten = _90X/5001,2,Set(CALLERID(name)=44578999)
exten = _90X,3,AGI(agi://127.0.0.1:4577/call_log)
exten =
(Message re-ordered and excessive quoting trimmed.)
On Tuesday 10 May 2011, mahesh katta wrote:
On Tue, May 10, 2011 at 2:00 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
I think it needs to be more like this. Here, I'm taking an educated
guess that you want your caller ID to appear
in the
archives.)
On Tuesday 10 May 2011, mahesh katta wrote:
On Tue, May 10, 2011 at 4:49 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
Not working can mean a lot of things.
So, let's start at the beginning. Have you ever actually managed to get
an outgoing call to work *at all* -- i.e
On Wednesday 11 May 2011, mahesh katta wrote:
Sir,
I set the below configured in Zapata.conf file. and A .J given Dialplan .
that's it is working now
hidecallerid=no
restrictcid=yes
Glad you got it all sorted -- I was going to suggest a few more things you
could try this morning, but got
This is beginning to turn even more unpleasant than the original breach of
netiquette which prompted the discussion -- like one of those fights which
starts with a raised voice, escalates to fisticuffs and then weapons, and by
the time innocent bystanders are getting injured nobody can even
On Tuesday 17 May 2011, Mike wrote:
Hi,
Is there any softphone or TAPI plug-in that allows one to dial from a web
page?
Just write a simple CGI script (running from the Asterisk server) which
looks up the nearest phone from the remote IP address ( $ENV{REMOTE_ADDR} in
Perl), and inject a
On Friday 20 May 2011, Dovid Bender wrote:
I had issue with call files. They would lock up the system (this was 5
years ago so maybe things have changed.)
Whenever you open a file for writing, a link is created in the containing
folder's directory (which says where on the disk the file is
On Friday 20 May 2011, salaheddine elharit wrote:
Ok thank you so much for all advice
This might help you a bit, too:
?php
$spool = /var/spool/asterisk/outgoing/; # outgoing callfile folder
$filename = asterisk- . date(U) . - . $_SERVER[REMOTE_PORT] . .call;
# this should end up being fairly
On Monday 23 May 2011, Thomas Perron wrote:
This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
but I want to know in any case!
Can a vb script run somehow on a Linux machine or does it only work on
Windows?
AFAIK there is no Linux interpreter for VBS :( But the
On Tuesday 24 May 2011, BroadTel wrote:
Hi all,
Just in case if anyone will be interested in *REDACTED*, a USB to FXS
adapter embedded with SIP softphone. Product specification is as follows:
Please refer to the fifth and sixth words of the title of this mailing list.
To everyone else, I
On Tuesday 24 May 2011, jon pounder wrote:
On 05/24/2011 11:35 AM, A J Stiles wrote:
Someone asked about the quality of it, he was quoting the hardware specs
of a similar device.
No they didn't. The original message to which the spammer was pretending to
reply (and in the wrong place
On Wednesday 25 May 2011, randulo wrote:
On Tue, May 24, 2011 at 10:50 PM, Matt Darnell mattdarn...@gmail.com
wrote:
We expect that users of Skype for Asterisk will be able to continue
using their Asterisk systems on the Skype network until at least July
26, 2013. Skype may extend this at
On Thursday 26 May 2011, virendra bhati wrote:
How to make outgoing calls from DID and what is theway to get incoming
calls from DID.
First of all, get your dialplan and zaptel configuration working to the extent
as you can make SIP to SIP calls between extensions, and you can make
outgoing
On Friday 27 May 2011, salaheddine elharit wrote:
i have installed asterisk and i have 3 sip 104 ,105 and 106
Now I can make the calls with theses sip without issue
I want to configure the outbound calls for these sips like that:
104 permission to call any number, but for 105 and 106 I want
On Friday 27 May 2011, satish patel wrote:
Hi There,
We have single PRI with multiple DID numbers and its working fine in
receiving call. And if you make outbound call it will send main-line
CallerID (company name). Now we want individual caller id for per
extensions on outbound calls. like
On Thursday 02 Jun 2011, khalid touati wrote:
Hi Guys,
Actually My question is as in the subject, may I use a regular phone line
to receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8.
Yes, you can. BUT, you will need some sort of FXO interface (allows the
computer to
On Friday 10 Jun 2011, mahesh katta wrote:
Hi,
I have 44578900 to 44578999 DID's. and I have extensions(100) for this
DID's. but problem is
callerid Extensions
44578900 100
44578901 101
44578902 102
44578902 103
44578903 104
44578905 200
44578906
On Friday 10 Jun 2011, Steve Totaro wrote:
Why do programmers try to make solution so elegant when an entries for
each phone in sip.conf is all that is needed.
No need for mathematical formulas, AGIs, and databases. You just took
over engineering to a new level.
Because doing it your way
On Friday 10 Jun 2011, Steve Totaro wrote:
I never understood hy people who have block of DIDs in a row choose to
make life difficult by not incrementing extensions by one, send caller
ID by prepending the common numbers and only sending four digits.
Well, to be fair, that's what most people
On Thursday 16 Jun 2011, mahesh katta wrote:
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any
On Thursday 16 Jun 2011, mahesh katta wrote:
-- Executing Set(Zap/3-1, Dest=50{EXTEN:-2}) in new
stack
-- Executing MixMonitor(Zap/3-1,
/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2
225.gsm|av(0)V(0)) in new stack
-- Executing Dial(Zap/3-1,
On Wednesday 29 Jun 2011, Abid Saleem wrote:
Hi All,
I have 100 Trunks from my Provider. My Provider is restricting me to make
only 120 minutes Call duration / trunk / day. So I want to load balance my
calls to these 100 trunks. Please advise in this regard ASAP. Thanks in
advance.
On Wednesday 29 Jun 2011, Marc Smith wrote:
Hi,
This is a bit of a long shot and I don't have much information on what
is actually happening...
Our production Asterisk system: ~2,000 SIP handsets, (2) Digium
TE220s, Asterisk 1.6.2.18, RHEL 5 x86_64
Every few weeks, or few months, or X
On Thursday 30 Jun 2011, michael k wrote:
All,
I am new in Asterisk. I am using asterisks with freepbx 2.7.0.10
version. I have tried to setup voice mail by dialing *97 from my extension.
The prerecorded system asking for a pond key at the end of each
recording. But unfortunately i
On Friday 01 Jul 2011, asterisk asterisk wrote:
I have this error after upgrading to 1.8.4.4 on my centos 5.6 32it
When using GUI to access, I got this error
*** glibc detected *** /usr/sbin/asterisk: double free or corruption
(!prev): 0x0919c070 ***
The server cannot be connected via GUI
On Friday 01 Jul 2011, Abid Saleem wrote:
Dear AJS,
Thank you for your response with good idea. Unfortunately I am not good at
programming. Can you please write this AGI script for me. Please help if
you can.
Sure I can help.
But you'll need to contact me off-list, as the rules here forbid
On Friday 08 Jul 2011, salaheddine elharit wrote:
i want to use timeout with asterisk 1.4 in order to hangup the outbound
calls after 25 sec
i call my mobile number 067xxx from my sip acount 223 and i want to
hangu up the call automatic after 25 sec but there is no hangup after 25
On Friday 08 Jul 2011, salaheddine elharit wrote:
what can i do in order to fix this issue
If and when an absolute timeout occurs, Asterisk jumps to the T extension.
So, in the same context as your 223 extension, you need something like
exten = T,1,NoOp(Absolute timeout triggered)
exten =
On Tuesday 12 Jul 2011, d tbsky wrote:
hi:
I am a SFA (skype for asterisk) user. I had ask Digium questions
about SFA usage in the future. but they seem too busy to reply. so I
tried at this list. I hope there are SFA users or Digium people can
solve my confusion.
Poor you!
To my mind,
On Thursday 14 Jul 2011, Kaushal Shriyan wrote:
Hi,
Any time line of availability of Asterisk binaries on CentOS version 6.
Yeah . as soon as someone compiles them :)
Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even
take long anymore (on any target system
On Wednesday 20 Jul 2011, Antonio Modesto wrote:
I am writing a Asterisk dialplan from scratch (for learning and
testing purposes), but i'm having trouble with a algorithm to dial a SIP
group using round-robin. I want that asterisk dial the member of the
group in a circular way, until the
On Tuesday 26 Jul 2011, Gilles wrote:
Hello,
Since Asterisk has been ported to exotic platforms like SOHO routers
(Linksys, Buffalo, etc.) and non-MMU CPUs (Blackfin, etc.), I was
wondering why the Windows port never really took off.
A better question would be: Why would anyone even *want*
On Tuesday 26 Jul 2011, Gilles wrote:
On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon)
soeren.malc...@mcon.net wrote:
And asterisk just runs fine on linux why bother ?
Because I, for one, would like to run Asterisk on my Windows
workstation at home as an enhanced answering machine
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