Le 20/09/2012 06:22, Fábio Lira a écrit :
Hello all,
I bought a Cisco model of SPA-301, I saw that in the specifications it supports
SRTP for secure connections, but I'm not finding how to create a certificate
and where and how should I configure my Asterisk 1.8. I already use other ip
Dear AJS;
I have fedora core 14 and I did yum install libsrtp-devel and it is already
existed, the only thing happened that it updated it.
Currently:
libsrtp-1.4.4-1.20101004cvs.fc14.i686
libsrtp-devel-1.4.4-1.20101004cvs.fc14.i686
Again, I did make menuselect and the same problem, I am not
Hi list,
in asterisk 1.4 and maybe earlier it was possible to use voicemail
system with mailboxes starting with some special characters like *. The
line in voicemail.conf was like this:
*123 = , AB,,,tz=cet|attach=no|
Calling exten = s,n,Voicemail(*123,su) is working in asterisk 1.4.
In
Hi all,
For one of my inverstigations it looks like i'm back to square one
I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail.
No google is involved as i use a local xmpp server (ejabberd)
I was experimenting on 1.8.15.1 (with jabber.conf,
this is quite complicated to be setup. however you can try using :
asterisk 1.4.11 with libpri patch for h324m and app_h324m.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
- Original Message -
From: Karsten Wemheuer k...@gmx.de
To: asterisk-users@lists.digium.com
Sent: Thursday, September 20, 2012 2:28:07 AM
Subject: [asterisk-users] Voicemail not working with vm boxes named with
astar
Hi list,
in asterisk 1.4 and maybe earlier it was
Hans Witvliet wrote:
Hi all,
Hola,
For one of my inverstigations it looks like i'm back to square one
I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail.
No google is involved as i use a local xmpp server (ejabberd)
I personally am using
- Original Message -
From: Marco Colombo mcolo...@enter.it
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 19, 2012 10:51:43 AM
Subject: [asterisk-users] SIP CANCEL, Reason
Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the
Hi list,
Am Donnerstag, den 20.09.2012, 09:28 +0200 schrieb Karsten Wemheuer:
Hi list,
in asterisk 1.4 and maybe earlier it was possible to use voicemail
system with mailboxes starting with some special characters like *. The
line in voicemail.conf was like this:
*123 = ,
Hi Matthew,
Am Donnerstag, den 20.09.2012, 06:27 -0500 schrieb Matthew Jordan:
- Original Message -
From: Karsten Wemheuer k...@gmx.de
To: asterisk-users@lists.digium.com
Sent: Thursday, September 20, 2012 2:28:07 AM
Subject: [asterisk-users] Voicemail not working with vm boxes
Has anyone ever run across where asterisk looses part of a diaplan???
I has this happen a couple times, so I put script in place at 2AM that dumps
the dial plan and compares it to the previous day or a know good one.
This ran fine for quite a while (multiple weeks, forget when I started
this).
Has anyone ever run across where asterisk looses part of a diaplan???
I've been running Asterisk since pre 1.0 days and to date, I've never had this
happen. My guess is you have some type of outside process modifying your
dialplan?
Either way, I have a BackupPC install that does nightly
Looks like you have Ghost doing mischief with your Asterisk. :P
On Thu, Sep 20, 2012 at 5:53 AM, Doug Lytle supp...@drdos.info wrote:
Has anyone ever run across where asterisk looses part of a diaplan???
I've been running Asterisk since pre 1.0 days and to date, I've never had
this happen.
On Thursday 20 September 2012, Jerry Geis wrote:
Has anyone ever run across where asterisk looses part of a diaplan???
I has this happen a couple times, so I put script in place at 2AM that
dumps the dial plan and compares it to the previous day or a know good
one. This ran fine for quite a
Unless you have configured your file systems not to, there will be a
modification
time on the extensions.conf. That might give you a clue as to *when* it got
altered.
--
AJS
The date is Aug 2 2012.
So the file is not changing.
Anything else?
Jerry
--
Any ideas on how asterisk could accept an email (such as an email to SMS or
num...@mybox.org sort of thing) and make a phone
call to a specific number and make an announcement?
I imagine the first part is the big question.
joe a.
--
Joseph Acquisto wrote:
Any ideas on how asterisk could accept an email (such as an email to SMS or
num...@mybox.org sort of thing) and make a phone
call to a specific number and make an announcement?
I imagine the first part is the big question.
Stuffing this logic into Asterisk really isn't
If you're using sendmail and receive e-mail directly to your box, you could
create a user and add a .forward file that pipes the e-mail to a script
which access the Asterisk Manager interface or something of the like.
There's lots of tutorials on both.
Good luck!
On Thu, Sep 20, 2012 at 12:31
Hello list.
I am facing a small problem when I try to run a macro that is in AEL
through extensions.conf. I'm by applying Macro () invoking the macro, but
it always generates this message:
[09.20.2012 10:43:23] WARNING [28923] app_macro.c: No such context
It may not be extensions.conf per se. It could be extensions-custom.conf or
any other file included in extensions.conf. Also, Asterisk generates some
of its' own custom context entries, so you might look into that as well.
Also check extensions.ael.
From:
It may not be extensions.conf per se. It could be extensions-custom.conf or
any other file included in extensions.conf. Also, Asterisk generates some
of its' own custom context entries, so you might look into that as well.
Also check extensions.ael.
Danny
THanks, actually all of my
Hi All,
I have a scenario where leader is giving a lecture and other participants are
on mute...
At the end of conf , when QA session begins is there a way for participants to
raise hands if they have questions so Leader can unmute them. Is this feature
already there in Meetme conf ? If there
Which Asterisk version?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pankaj pandey
Sent: Thursday, September 20, 2012 2:12 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Hand Raise|Meetme Conf
Hi All,
I have
thanks for your prompt response .
I am using asterisk 1.4.32 ,please suggest which version i have to use.
Thanks Regards,
Pankaj Pandey
+91-9990212758
From: Danny Nicholas da...@debsinc.com
To: 'pankaj pandey' pankaj.n...@yahoo.com; 'Asterisk Users Mailing
I've been working on an interactive XMPP interface so users at my office can
interact with the timeclock and queues by XMPP (in addition to IVR menu, which
has been running just fine for quite a while before the XMPP interface). I'm
using sendtodialplan=yes to handling the incoming unsolicited
On Thu, 20 Sep 2012, Jerry Geis wrote:
Actually I restart asterisk every day at 2AM. So something happens in a
24hour window.
On Thu, 20 Sep 2012, Jerry Geis wrote:
THanks, actually all of my modifcations were to the extensions.conf file
itself. It seems like those are the ones that got
On Thu, 20 Sep 2012, Joseph Acquisto wrote:
Any ideas on how asterisk could accept an email (such as an email to SMS
or num...@mybox.org sort of thing) and make a phone call to a specific
number and make an announcement?
I imagine the first part is the big question.
procmail could be a
- Original Message -
On Thu, 20 Sep 2012, Jerry Geis wrote:
Actually I restart asterisk every day at 2AM. So something happens
in a
24hour window.
On Thu, 20 Sep 2012, Jerry Geis wrote:
THanks, actually all of my modifcations were to the extensions.conf
file
itself. It
The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 11.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users of Asterisk are encouraged to participate in the
Asterisk 11
At 09:40 AM 9/20/2012, you wrote:
All interested users of Asterisk are encouraged to participate in
the Asterisk 11 testing process.
Is this the equivalent of that?
svn checkout http://svn.asterisk.org/svn/asterisk/trunk
I realize it's different, but I'm curious if it's at least that or is
Ira wrote:
At 09:40 AM 9/20/2012, you wrote:
All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testing process.
Is this the equivalent of that?
svn checkout http://svn.asterisk.org/svn/asterisk/trunk
I realize it's different, but I'm curious if it's at least
On 20/09/2012, at 3:59 PM, Noah Engelberth n...@directlinkcomputers.com wrote:
I’ve been working on an interactive XMPP interface so users at my office can
interact with the timeclock and queues by XMPP (in addition to IVR menu,
which has been running just fine for quite a while before the
On 20/09/2012, at 4:07 PM, Steve Edwards asterisk@sedwards.com wrote:
On Thu, 20 Sep 2012, Joseph Acquisto wrote:
Any ideas on how asterisk could accept an email (such as an email to SMS or
num...@mybox.org sort of thing) and make a phone call to a specific number
and make an
On 20/09/2012, at 3:59 PM, Noah Engelberth
n...@directlinkcomputers.commailto:n...@directlinkcomputers.com wrote:
I've been working on an interactive XMPP interface so users at my office can
interact with the timeclock and queues by XMPP (in addition to IVR menu, which
has been running just
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