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
Hi,
Do these IMSI names / numbers match what your phone is trying to register
as? Are there actual at the end of the numbers, or are you
attempting to obfuscate?
yes xxx are numbers (not real letters x), it's just 'obfuscation' and
anyway it's easier to recognize them by the first few
On Thu, 2011-02-24 at 23:24 -0300, Andrew Latham wrote:
On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Has this issue been fixed in this release of 1.8 (or even in the
previous 1.8.2.3)?
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
On Thu, 24 Feb 2011 13:11:02 + (UTC), t...@mountifield.org (Tony
Mountifield) wrote:
Yes, that is the reason. The easiest thing is probably to put in a delay
if os.execute allows full shell syntax:
Thanks for the idea.
While reading samples, I happened upon the system() application,
which
On Fri, Feb 25, 2011 at 4:54 AM, Gilles codecompl...@free.fr wrote:
Is there a way to launch a script asynchronously, so that Asterisk
proceeds to the next step immediately, and the script will then wait
10 seconds so that the channel is available again?
In Perl, the line would be: fork and
On Fri, 25 Feb 2011 05:47:36 -0600, Tilghman Lesher
tilgh...@meg.abyt.es wrote:
I'm sure there's an equivalent in lua, but the basic idea is that you
want to fork a
child, which takes over. When the parent process dies, control returns to the
dialplan.
Good idea. uClinux supports vfork() instead
In article a92fm65o07976iomfhr1k0t3sunmogd...@4ax.com,
Gilles codecompl...@free.fr wrote:
On Thu, 24 Feb 2011 13:11:02 + (UTC), t...@mountifield.org (Tony
Mountifield) wrote:
Yes, that is the reason. The easiest thing is probably to put in a delay
if os.execute allows full shell syntax:
Hi folks,
How handle in dialplan when a user disconnected ?
The first try was using the special context h, however all hangups are handled
by this context and I only need identify when user's hangup in a
Read/Background/Dial application.
Can asterisk send to dialplan only when the user
On 11-02-24 08:56 PM, Andrew Latham wrote:
And I go back to triple check and compare revision numbers... You are
100% correct, the revision numbers in our local repository are wrong,
someone pushed the 1.8.3 RC3 into our local 1.8.2 branch. I apologize
and will work to better control my trust
On Fri, Feb 25, 2011 at 9:49 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 11-02-24 08:56 PM, Andrew Latham wrote:
And I go back to triple check and compare revision numbers... You are
100% correct, the revision numbers in our local repository are wrong,
someone pushed the 1.8.3 RC3
Sidarta Aguiar de Oliveira wrote:
only need identify when user's hangup in a Read/Background/Dial
application.
Set a flag for those cases and then trap it in the H extension
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
On Fri, 25 Feb 2011 12:20:34 + (UTC), t...@mountifield.org (Tony
Mountifield) wrote:
os.execute((sleep 2;mv /var/tmp/callback.call /var/tmp/asterisk/outgoing))
The important parts are the ( before the sleep and the ) at the end. The
brackets create a subshell to do the sleep and move, and the
Hi ben,
actually the ${OUTBOUND} is generated by the php... but based on what
will be received on ${ARG1}. unfortunately i am not getting the value
from the argument. not sure why. thanks again.
regards
Ron
On Friday 25,February,2011 11:10 AM, Ben Klang wrote:
On Feb 24, 2011, at 5:27
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
Sent: Friday, February 25, 2011 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] missing argument on AGI
Hi
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron
actually the ${OUTBOUND} is generated by the php... but based on what
will be received on ${ARG1}. unfortunately i am not getting the value
from the argument. not sure why. thanks again.
On Fri, 25 Feb 2011, Danny Nicholas
Hi,
I'm wondering if this is normal asterisk behaviour:
asterisk*CLI sip show channels
Peer User/ANR Call ID Format Hold
Last MessageExpiry Peer
10.12.0.2(None) 3c2f7ff2975e-wp 0x0 (nothing)No Rx:
PUBLISH
Hello all,
After numerous issues with Snom phones (360/370/870) potentially looking to
migrate too Yealink as their product range looks very promising indeed.
Are any of you using them with Asterisk ? How do they perform ? Do you use mass
deployment at all ?
Would be very interested to
I am an Italian manufacturer of PBX system
Before using snom phones but when I tried the phones Yealink I no longer
changed.
and 'a nice phone and very good, like customers
I using this phone with asterisk 1.6.2.15 i don't have a problem.
I using a provisioning an its very simply to
Hello Doug,
I've tried to use the 'hangupcause' variable like flag but in the both of cases
(when asterisk hangs up and the called party hangs up) the variable have the
value 16 (normal clearing). Theres another variable that can i use for this
purpose?
Luiz Gustavo Chiaretto
From:
Luiz Gustavo Chiaretto wrote:
only need identify when user's hangup in a Read/Background/Dial
application.
If you're going to be doing a read/background/dial, then just before any
of them, set a variable. For example:
exten s,1,Set(_Flag=Read)
exten
In article 4q9fm6h53jk91la6nbtvhhe4cfah89h...@4ax.com,
Gilles codecompl...@free.fr wrote:
Does it mean that there's no way for Zaptel to know the status of a
call (ring, answered, busy, etc.), and the only way is to use Wait()
and hope for the best?
If you are using an analogue phone line, I
On our live server, running Asterisk 1.4.23.1 and DAHDI-Linux 2.1.0.4. On
occasion (not too rare, happens maybe once every month or two), the PRI
and/or DAHDI will stop working properly and we'll get repeated messages
like this:
[Feb 25 05:17:22] VERBOSE[9511] logger.c: -- B-channel 0/1
On Fri, Feb 25, 2011 at 3:01 AM, Axelle aaforti...@gmail.com wrote:
yes xxx are numbers (not real letters x), it's just 'obfuscation' and
anyway it's easier to recognize them by the first few digits.
and yes, they match the phone.
Show us. The error you're receiving specifically states that
Thanks a lot for the idea Doug but it's complicated to use because i have to
unset the flag every time that asterisk hangs up the call. Somebody knows
another solution for our problem.
Thanks !
Luiz Gustavo Chiaretto
- Original Message -
From: Doug Lytle supp...@drdos.info
To:
i installed skype for asterisk
i can send and recieve calls normaly
how can i receive messages from another skype user
i Succeed to send only
using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message
text)
how to receive messages using this code
Maybe something like this?
[skype_chat_receieve]
Exten = account,user,1,do something here?
What do you see in the CLI on the incoming txt message?
I just figured out how to handle a different google talk account today
[google-in]
Exten =
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
any idea
regards
Khaled
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received
Can you please send me a how to please or a simple lines?
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: mailto:kche...@xplorium.com
I am assuming that goes the same for Gtalk chat messages too?
Or has nobody played with that?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing
AFAIK, the issue here is not Skype or Gtalk. The Asterisk client isn't
really designed to easily transport messages during the call or otherwise.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent:
Hey guys,
We have asterisk 1.2.7.1 running and today i got following error message. and
users started complain regarding call issue. after reboot everything comes back
to normal. I just want to know what happened ?
Feb 25 11:27:11 WARNING[10042] app_meetme.c: Error setting conference
Feb 25
Can you please send me a how to please or a simple lines?
Regards
Please see the README file that came with skypeforaterisk. Search for
SkypeChatMessage.
As far as AMI tutorial, please see Asterisk: The Definitive Guide chapter 20
(and consider ordering a copy).
On Fri, 25 Feb 2011 17:31:42 + (UTC), t...@mountifield.org (Tony
Mountifield) wrote:
If you are using an analogue phone line, I believe that's true.
Much better if you can to use either a digital phone line (ISDN)
or else a VoIP connection. Either of those will be able to tell
properly when
Using a Digium T1 module (TE420 and others). Trying to keep specific channel
audio always enabled regardless of signaling states. The channels
are configured as em. I still require the em signaling for external
control (so can't use bchan or inclear), but do not want them to control
the audio
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