No tell me that's a jock ! I can't believe it:
http://nerdvittles.com/?p=7940
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi,
I have trouble establishing a call between between two SIP phones. One sip
phone is, with asterisk server, at home behind a firewall. The second sip
phone is an iPhone with 3G wireless connection.
When I call from the SIP device at home the SIP account on the Internet
(iphone + 3G) I can h
To Jonas:
I have an asterisk box at home and I have this line in my rtp.conf file:
rtpstart=1
rtpend=10100
And My FW is setup to forward all incoming ports of range 1-10100 to
the asterisk PC.
I've never had a problem since one year, but I have never received more
than two simultaneous
Hi,
I want to recieve calls to my Skype account and forward them to a SIP/FXS
line. I searched for chan_skype for asterisk (v11), but found it only
available for asterisk 10
I know that Digium gives no support for this module, but I am sure that
someone somewhere did write some tool to allow such
Hi,
I don't know how to call this functionality, but what I want to do is join
an already established communication between PSTN---FXS_connected_phone
using my SIP phone (I have an asterisk v11 with digium TDM400P at home)
Is it possible? What I don't want is using the conference sound and
menu..
Hi,
I am using asterisk 11.1.0. How to display the caller number (from asterisk
-rvvv terminal) in the first step of the extension (before doing any
action) ?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-dig
I have a recurrent problem on my asterisk box. I have "VIA Samuel 2" as a
CPU. With asterisk v11.1.0 and dahdi-linux-complete-2.6.1+2.6.1 compiled
from source.
I get a RED alrm drom the port 1( FXO) two or three times per day:
[Feb 4 15:54:57] WARNING[9991]: chan_dahdi.c:8018 handle_alarms: Det
uld I contact debian dev team for that?
Thanks
OLD messages ---
Message: 6
Date: Mon, 31 Dec 2012 12:08:40 -0500
From: neo haux
Subject: Re: [asterisk-users] Compile asterisk11.1 for i586 VIA C3 CPU
To: asterisk-users@lists.digium.com
Message-ID:
Content-Type: text/plain; charse
="utf-8"
Try this... In menuselect, uncheck BUILD_NATIVE under Compiler Flags and
recompile.
On Sun, Dec 30, 2012 at 4:44 PM, neo haux wrote:
> Hi,
>
> I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2
> 800MHz CPU. A small box to play with PBX at home
Hi,
I've compiled asterisk 11.1 for my MiniITX card with VIA C3 Samuel2
800MHz CPU. A small box to play with PBX at home.
I get this error when I start asterisk:
root@pbx01:/usr/src/asterisk-11.1.0# /etc/init.d/asterisk start
Illegal instruction
Starting Asterisk PBX: asteriskIllegal instructio
Commercial Discussion
Message-ID:
Content-Type: text/plain; charset=ISO-8859-1
On Sat, Jun 23, 2012 at 10:32 AM, neo haux wrote:
> Actually I can start and receive SIP calls (PC client, iphone client)
> but?I have an issue with calling external number throught PSTN
> (c
Actually I can start and receive SIP calls (PC client, iphone client)
but I have an issue with calling external number throught PSTN
(certified-asterisk-1.8.11-cert2).
I'm having this error when making a call:
*CLI> == Using SIP RTP CoS mark 5
-- Executing [9000@local:1] Dial("SIP/3000-000
ving in Canada, so I guess I should use USA signaling ? If so in
which file ?
Message: 7
Date: Tue, 04 Oct 2011 14:49:55 -0400
From: John Novack
Subject: Re: [asterisk-users] Delay before ringing from PSTN`s call
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Hi
I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
I configured incoming calls from pstn to ring my SIP phone (extension : 100)
cat extensions.conf
...
[from-pstn]
exten => s,1,Dial(SIP/100,10)
same => n,VoiceMail(100,u)
root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
...
..
Hi rcswebb,
I had a problem like yours :
Asterisk -NAT - internet - NAT - 3CX phone
Without modifiyng Astrisk conf I could start a call from the client but
without hearing a sound.
The solution for me was to force Asterisk to modify the outgoing udp packet
to insert it's public ip and not the p
Mailing List - Non-Commercial Discussion
Message-ID: <0d19da7a-0766-4ae9-a967-528ccae36...@gmail.com>
Content-Type: text/plain; charset="us-ascii"
what do you mean? Like speed dial or directory?
Sent from my iPhone
On Sep 4, 2011, at 6:47 PM, neo haux wrote:
> Hi,
&g
Hi,
It is possible to save all the phones numbers on asterisk servers instead of
doing so manually in each VoIP device ?
Does SIP take care of such configuration ?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.a
Hi
I want to change my old answering phone machine and two wireless phones with
asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel
9133i) + Wifi/SIP phone
I am wondering if I´ll lost actual functionalities that are present in my
old answering machine:
1) is it possible to sho
AP
On Aug 2, 2011, at 12:06 PM, neo haux wrote:
> Hi,
>
> I?ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy)
> I also compiled iksemel (v1.4) with the option 2./configure
> --with-libgnutls-prefix=/usr"
> As explained in this link (to avoid compilation error
Hi,
I´ve compiled asterisk-1.8.5.0 on my Debian based distro (Pinguy)
I also compiled iksemel (v1.4) with the option 2./configure
--with-libgnutls-prefix=/usr"
As explained in this link (to avoid compilation error )
http://code.google.com/p/iksemel/issues/detail?id=29#c3
I configured jabber.con
20 matches
Mail list logo
