Hi,
I think so yes unless somebody else can provide a better solution. (Perhaps I'm
doing it wrong ;-) )
We have 2 asterisk servers (Xivo distribution based on Debian) whom work in
Active/Passive cluster mode. Then we have a third server which is the OpenFire
server (based on Ubuntu 14)
So
You can have the openfire server installed on the same server as asterisk
without any issue, just size your server appropriately. Just keep in mind
they are different services.
James Cass http://goog_987864563
jcas...@gmail.com
On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com
I just get started with it so my question maybe not well catch. Anyway to
do the VOIP call and IM we need to use two difference servers? which one is
asterisk for VOIP ? and other one for IM that is openfire ? or we can have
other choice better than this ?
Thank you for your help, I am waiting for
On 6 July 2015 at 15:27, Motty Cruz motty.c...@gmail.com wrote:
Hello,
I would like to setup a mechanism to trigger an alarm if user is deal too
many numbers within a very short period of time. Safeguard against users
hacked accounts.
can someone help?
Thanks,
You could use fail2ban
Indeed, thanks.
I'll let you know how it goes.
Thanks,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 22:24
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
You could use the group functionCreate the group by extension and check how many calls are in the groupIf it's more than you allow then have it send a email
Hi list,
we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual
running version. Patch failed with
zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0
../asterisk-11.18.0-patch
can't find file to patch at input line 5
Perhaps you used the wrong -p or --strip option?
The
Rodrigo Pimenta Carvalho wrote:
Hi.
In my dial plan I can use the following commands to access and handle
data from the builtin database.
DB DB_DELETE DB_EXISTS DB_KEYS
Equivalents exist for these in the Asterisk Manager Interfaces as
actions. An example being DBGet[1] but others also
Richard Kenner wrote:
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f $6 = {frametype = AST_FRAME_VOICE, subclass = {integer
= 100021, format = {
Rodrigo Pimenta Carvalho wrote:
Hi.
I have a beginner conceptual question about Asterisk:
Let's suppose that there are 4 softphones registered in my Asterisk
and all of them are currently online. In addiction , there is no
call.
Suddenly, one of these softphones sends a SIP message to the
On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI ad...@tootai.net
wrote:
Hi list,
we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual
running version. Patch failed with
zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0
../asterisk-11.18.0-patch
can't find
Hi,
I am new to asterisk, I have set up the asterisk server and successfully I
could make the dialplan between 2 SIPs but when there are more than two
sips calling each other, my dialplan seems doing the wrong routing to the
sip. Do i need to config anything additionally to asterisk to handle
On Wednesday 08 Jul 2015, Thyda ENG wrote:
Hi,
I am new to asterisk, I have set up the asterisk server and successfully I
could make the dialplan between 2 SIPs but when there are more than two
sips calling each other, my dialplan seems doing the wrong routing to the
sip. Do i need to
Yes, I have though of setting them up on the same server(openfire, and
asterisk) and the problem come in mind that how can register the user to
openfire automatically when I register the user SIP on the asterisk server
? Do you have any idea? I am waiting for your reply.
Thank,
Thyda
On Wed,
The asterisk plugin for openfire would be what I would think would do that,
but as another person posted, it's very deprecated, so I'm not sure how
well it would work. I've never used it personally.
James Cass http://goog_987864563
jcas...@gmail.com
On Wed, Jul 8, 2015 at 11:45 AM, Thyda ENG
This is an interpolated frame from func_jitterbuffer. It's part of
packet loss concealment. What scenario exposed this?
We were testing for clipping by doing Set(VOLUME(RX)=100) but we were
connecting to a ConfBridge that had a jitterbuffer. This occurred when
the phone (SIP) hung up.
--
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls
Hi Joshua Colp.
Thank you very much for alerting me about the impossibility of forwarding the
SIP 183 messages from callees to caller, via Asterisk, when more than 1 callee
ring at same time.
In my project the caller software (a proprietary softphone) needs to know some
information about the
2015-07-08 13:11 GMT-06:00 Joshua Colp jc...@digium.com:
You probably want to add rewrite_contact=yes to your endpoint. This will
cause it to reuse the existing connection established from the phone.
Generally the port provided by the phone is not reachable.
Hi Joshua , I add the option you
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
I'm trying to receive fax from PSTN, with the following setup:
Fax machine --- PSTN --- *11 --- *13 --- IAXmodem + Hylafax
Fax machine is connected to the PSTN, call arrives via ISDN on Asterisk
11.16.0 used as gateway, chan_sip relays the
Le 08/07/2015 17:36, Richard Mudgett a écrit :
On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI ad...@tootai.net
mailto:ad...@tootai.net wrote:
Hi list,
we wanted to patch our servers with 11.18.0 patch against 11.17.0
actual running version. Patch failed with
2015-07-08 13:09 GMT-06:00 Ryan, Travis ry...@oscarwinski.com:
Asterisk13 can do native tls with each phone? Nice.
any example?
rickygm
http://gnuforever.homelinux.com
--
_
-- Bandwidth and Colocation Provided by
Hi Thyda
When you set exten = _.,1,Dial(SIP/${EXTEN}) Asterisk assume _., an match
everything on your dialplan including special extensions as i, h, etc.,
these will cause problems onto your system.
If you need to match one or more digits you can use _x and _x.
_x it mean only one pattern
ricky gutierrez wrote:
Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed
to make it work, all my terminals spa Cisco 5XX
look my cli
[Jul 8 11:09:16] ERROR[14733]: pjsip:0?:tlsc0x7f539801 TLS
connect() error: Connection refused [code=120111]
[Jul 8 11:09:16]
Hi list , I'm doing some tests with asterisk 13.4 and tls, and failed
to make it work, all my terminals spa Cisco 5XX
look my cli
[Jul 8 11:09:16] ERROR[14733]: pjsip:0 ?:tlsc0x7f539801 TLS
connect() error: Connection refused [code=120111]
[Jul 8 11:09:16] WARNING[14733]: pjsip:0 ?:
Asterisk13 can do native tls with each phone? Nice.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ricky gutierrez
Sent: Wednesday, July 08, 2015 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial
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