On Monday 26 June 2017 at 14:06:10, Harel wrote:
> Hello List,
> I'm working on an autodialer project.
> At the moment I use the Originate application then I "throw" it to an
> extension where I Dial() the other party and then both legs are bridged.
>
> The problem is that the Dial() will only
On Monday 19 June 2017 at 18:12:35, Sebastian Gutierrez wrote:
> use replication
1. Agreed - use replication.
2. If you want an HA (High Availability, not dependent on a single Master DB
server replicating to a slave) solution, consider setting up Master-Master
replication, with an LVS (Linux
On Tuesday 06 June 2017 17:54:59 Mike Diehl wrote:
> Hi all,
>
> I'm upgrading to Asterisk 13.14.0 x86_64. During my beta testing, I've
> discovered that my server crashes as soon as I leave a voicemail message.
> I'm using odbc voicemail storage as well as mysql dynamic configuration.
>
>
On Tuesday 06 June 2017 16:57:07 andre castro wrote:
> On 06/06/2017 04:36 PM, Antony Stone wrote:
> >
> > Tell us about your networking arrangement - are both phones and the
> > Asterisk machine on the same network?
>
> Nop. They are in 2 different
On Tuesday 06 June 2017 15:18:32 andre castro wrote:
> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.
> But I hear nothing at the peer's end.
>
> When one peer calls another, sound comes through just fine.
Tell us about your
On Tuesday 23 May 2017 at 20:01:14, Tech Support wrote:
> Ok, the purpose of the answering machine detection (AMD) is to
> determine when the audio file should start playing *after* the call has
> been picked up. Typically, if a call has been picked up by a person, they
> say a short
On Tuesday 23 May 2017 at 19:20:25, Tech Support wrote:
> All;
>
> What I did was add a line in the dialplan that used the SendDTMF()
> application and that worked great. The problem that I’ve run into now is
> that dialing the extension screwed up the answering machine detection. The
>
On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote:
> Hello,
> I need to have an extension on a SwitchVox server dial out to one on an
> Asterisk (FreePBX actually) box which will host a voice directory.
What's a voice directory?
> The Asterisk server will then need to dial one of the
On Monday 08 May 2017 at 15:44:47, Marcelo Terres wrote:
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_Dialpl
> anExtensionAdd
>
> Is it enough?
Is there a similar call to delete an extension, or to modify an existing one?
On the basis that the OP already has extension
On Saturday 06 May 2017 at 09:21:16, Luca Bertoncello wrote:
> Antony Stone schrieb:
>
> > 4. Did the IP address of Telekom's end of the connection change?
>
> I really don't know, but I suppose not
I suspect this may in fact have been the cause of your problem.
Firstly, I
On Saturday 06 May 2017 at 08:37:39, Luca Bertoncello wrote:
> Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't
> connect to the remote Server (by Telekom) until today about 7:30.
>
> Well, it could happen...
> What I find really annoying was that I needed to restart
On Friday 05 May 2017 at 16:52:39, Richard Kenner wrote:
> > Of course, you might run into problems if the later release introduces
> > new options (or deprecates old ones) which then aren't going to be in
> > your makeopts file
>
> That's my question: how do I reflect the changes that I made to
On Friday 05 May 2017 at 16:21:20, Richard Kenner wrote:
> I'd like to be able to save the choices made in menuselect in a way
> that they can be tracked in a CM system and applied to a later release
> of Asterisk using an automated tool like Ansible. What's the best
> way to do that?
On Saturday 22 April 2017 at 22:25:52, Atux Atux wrote:
> Thanks a lot for the reply.
> I did follow that already, but i do have a problem. Here is my
> extensions.conf part for that particular number
> exten => 6912345678,1,Answer()
> exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root
On Thursday 20 April 2017 at 21:29:58, Steve Edwards wrote:
> Not an Asterisk question, but...
>
> A bunch of our 8xx numbers started playing this recording when dialed. Our
> provider (Inteliquent) says it's not them.
Where are Inteliquent feeding the calls (assuming they connect instead of
On Thursday 20 April 2017 at 18:31:03, Atux Atux wrote:
> root@PBX: /var/www/html $ /etc/init.d/asterisk start
> [ ok ] Starting asterisk (via systemctl): asterisk.service.
I'm somewhat puzzled that your root-user prompt is "$"
instead of the more normal "#", but never mind...
> root@PBX:
On Thursday 20 April 2017 at 12:31:14, Atux Atux wrote:
> Hi. thanks a lot for your replies. I did stop the services and i did issued
> the the "chown" and "chmod" commands listed in the guide.
> It is necessary to compile it, instead if using the apt-get version
> What am i missing?
Let's go
On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote:
> On 4/19/17 4:23 PM, Antony Stone wrote:
> >
> > You say the USB ethernet adapter got unplugged and then reconnected...
> >
> > 1. What's the name of the network device for this adapter? Is it the
> &g
On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote:
> On 4/19/17 4:09 PM, Antony Stone wrote:
> > On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:
> >>I have a server that had been operating for a few years now with
> >>
> >>
On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote:
> I have a server that had been operating for a few years now with
> IAX2 trunks to several other servers. Since yesterday all IAX2 trunks
> now say UNREACHABLE.
...snip...
> So far the only thing different is that the receive
On Wednesday 19 April 2017 at 18:48:29, Atux Atux wrote:
> Hi.
> Here is the output of the command
>
> root@pbx: ~ $ find / -name asterisk -exec ls -ld '{}' \;
>
> drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk
>
> drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32
On Wednesday 19 April 2017 at 15:44:39, Atux Atux wrote:
> hello there. i am running debian 8 in my swerver and i would like to run
> asterisk as non root. i did follow the
> https://www.voip-info.org/wiki-Asterisk+non-root without any success.
Did you do the very first step:
call: Conflicting extension values given. Using '832+ios' and not
'0203yyy'
== Using SIP RTP CoS mark 5
-- Called SIP/832+ios:31onpmlq_9d_x...@remote.server.com/0203yyy
[2017-02-28 11:43:47] NOTICE[11692][C-0d22]: chan_sip.c:23010
handle_response_invite: Failed to authenticate on
On Friday 02 March 2007 07:46, Alan Chandler wrote:
On Thursday 01 March 2007 20:33, bails wrote:
plug it in a linux box and tell us what it is please,
generic-usb-audio or what?
Bails
Julian Lyndon-Smith wrote:
Yeah, that's where firefly comes from, doesn't it.
I've got the
On Wednesday 21 March 2007 11:57, bails wrote:
Antony Stone wrote:
On Friday 02 March 2007 07:46, Alan Chandler wrote:
On Thursday 01 March 2007 20:33, bails wrote:
plug it in a linux box and tell us what it is please,
generic-usb-audio or what?
Bails
Julian Lyndon-Smith wrote
On Wednesday 21 March 2007 15:11, asterisk wrote:
I use this driver for the SJ phone with the USB tesco internet phone:
http://www.sjphone.org/usbphone/SJphoneDriverWebPoint.exe
Yes, but that's a corded phone which plugs into the USB socket.
# cat /proc/bus/usb/devices
P: Vendor=19af
On Wednesday 21 March 2007 19:54, Lutgring, Sam wrote:
Does anyone know how to configure a SIP phone to pass the mailbox number
to the voicemail service when dialing? I would like to press the
message waiting lamp and be prompted for my password instead of mailbox
number. Can this be passed
On Sunday 19 December 2004 20:18, Brian West wrote:
The SMS in asterisk is not SMS like you're thinking... Its not for sending
to mobile phones and not something usable in the US.
Um, sorry, but if SMS is not for sending to mobile phones, then what is it for
(if it matters, I'm not in the US)
On Sunday 19 December 2004 21:35, Antony Stone wrote:
On Sunday 19 December 2004 20:18, Brian West wrote:
The SMS in asterisk is not SMS like you're thinking... Its not for
sending to mobile phones and not something usable in the US.
Um, sorry, but if SMS is not for sending to mobile
On Saturday 18 December 2004 10:17, Norman Zhang wrote:
Hi,
May I ask what ports are necessary for SIP communication through a
firewall? I read somewhere that UDP/5060 alone is enough. Some
recommends more ports to be opened for RTP.
Both the above statements are correct.
SIP uses port
On Saturday 18 December 2004 10:58, Norman Zhang wrote:
SIP uses port 5060
RTP uses multiple ports, typically in the range 1-2
Remember that SIP and RTP are different - SIP is used to set up the call;
RTP is used to carry the audio once the call has been set up.
Thanks. May
On Saturday 18 December 2004 11:40, Rich Adamson wrote:
But, to return to my initial question, what's the security risk in
leaving your Asterisk server open to UDP packets from the world?
I regard it like a mail server - a firewall allowing TCP packets through
to port 25 cannot protect
On Saturday 18 December 2004 13:21, Tom Ivar Helbekkmo wrote:
My home firewall allows my Asterisk PBX to send any UDP traffic to
anyone, and keeps state, so they can answer. It also specifically
allows anyone to connect to UDP port 5060 on the PBX.
Interesting. Does that allow other people
On Saturday 18 December 2004 18:07, Dorn Hetzel wrote:
I wouldn't say I hate SIP, it sucks less than H.323 and
so on by a large margin. But, having said that, if you
can use IAX, it sucks even much than SIP does :)
Um, are you saying IAX is good, or that it is not good? I'm not sure I
On Saturday 18 December 2004 20:19, Bill Seddon wrote:
Detecting the ringing state of a specific device from, say, a desktop
running Windows or Linux AGI is trivial.
Care to share a trivial example with us?
Sounds like a useful link for several applications...
Antony.
--
Software
On Saturday 18 December 2004 20:27, Rodolfo Grave wrote:
Hi and thanks once more.
I moved the card around, and it kept the same IRQ. Then I went into
setup and changed it. This is the output of lspci -v now:
01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537
On Sunday 19 December 2004 00:26, Nabeel Jafferali wrote:
I have heard many times that IAX is NAT-transperant. I am unsure how
it accomplishes this.
I do know that SIP works like this: your SIP device send a request to
the SIP server (usually on port 5060) with whatever command. The SIP
On Sunday 19 December 2004 02:00, Keith O'Brien wrote:
Since the incoming stream is using VAD, my assumption is that it is losing
the timing during the pauses in the speech. Does anyone know of a way to
just turn off VAD in *? This would have multiple benefits (if you have
the bandwidth).
On Sunday 19 December 2004 01:41, Nabeel Jafferali wrote:
Thanks for all the info so far!
Therefore a NAT device between two IAX systems has only a
single channel, on a well-known port number, to deal with,
and this is simple to do.
So then how does IAX deal with the equivalent of SIP
On Friday 17 December 2004 14:31, Steven Kalcevich (Lists) wrote:
Why not just dial an extention for music when the user wants music
from there desk.
Because then the phone will be engaged on a call and will not ring when
someone else wants to talk to the person at the desk?
Antony.
The
On Friday 17 December 2004 16:22, Lee Howard wrote:
On 2004.12.17 05:42 Sergio Serrano wrote:
Hi all,
again I try configure Hylafax with asterisk. I would like
configure
Asterisk in the next way:
1)An incoming fax go into through X100P
2)Asterisk detects Fax and
On Friday 17 December 2004 18:34, Geoffroy KOUMADI wrote:
i have problem to setup application meetme. i'm using asteisk-1.0.3 and
sjphone as client.
Thanks for letting us know.
If you want some help in solving the problem, perhaps you might tell us what
the actual problem is?
Useful
On Friday 17 December 2004 20:24, Ferguson, Michael wrote:
G'Day All,
How do I fix this:
I receive a call at the extension. Press the hold button. Music on hold
starts. When I place the handset back on the cradle, the call gets hung
up/disconnected. The Phone is A GrandStream Budge Tone
On Friday 17 December 2004 20:43, Ferguson, Michael wrote:
OK. I guess I was not clear. Sorry.
The phone rings.
The person picks up the handset and speaks to the caller.
He then puts the call on hold by pressing the HOLD button on the GS
100 phone.
The caller hears music on hold.
So far,
On Friday 17 December 2004 21:00, Patrick Campbell wrote:
I am looking to help out my company find a more budget conscious but
reliable way to hold conference calls between 5+ people. 4x a month we
hold several hour long conference calls during non-business hours. All of
the employees have
On Friday 17 December 2004 21:10, Ferguson, Michael wrote:
Antony,
Thanks. It seems that the GS will not keep the call on hold.
In the real world though, when you place a call on hold, it is held until
further action.
Yes, although I might think that hanging up is a further action?
The
On Friday 17 December 2004 21:25, Ross Kevlin wrote:
this would still only work if the mailbox number was the same as the caller
id. I need some way to get the actual mailbox number of the caller.
Where / how are your mailbox numbers stored?
It shouldn't be too difficult to create a script or
On Friday 17 December 2004 21:42, Nihal wrote:
Does some hardware just not work very well with Asterisk?
Yes. (or, no, depending on how you view the question)
I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram.
Some people have reported problems with FC3, I don't know if FC2
On Friday 17 December 2004 23:04, Patrick Campbell wrote:
Come to think of it since the DTA310 uses DNS to find the SIP server, you
could setup a DNS cache and override the DNS entry for what packet8 uses
(proxy2-eqix-sjo.packet8.net : 15062, over here) to point to the IP of your
own SIP
On Thursday 16 December 2004 22:57, Jared Armstrong wrote:
I found IP 500's for $170.
Where?
Antony
--
The truth is rarely pure, and never simple.
- Oscar Wilde
Please reply to the list;
On Wednesday 15 December 2004 14:21, Roy Sigurd Karlsbakk wrote:
hi
using the following in sip.conf, codec preferences aren't set, and
asterisk uses alaw whatever I do, except force it to one specific in
the [user]
[general]
disallow=all
allow=g726
allow=g729
allow=gsm
allow=alaw
On Wednesday 15 December 2004 22:34, Carey Pillar wrote:
I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with
what seems to be correct settings (according to digium and asterisk wiki).
As soon as I plug in my POTS line into FXO mod the line goes into offhook
state (whether
On Thursday 16 December 2004 01:09, Shahed wrote:
Hello All,
I am new to *, and this is my first post on the user list.
I have had success with making / receiving calls to a SIP hardware Phone
and the Console Channel Driver.
Can anyone please suggest what would be a good SIP server to use,
On Wednesday 15 December 2004 21:26, Michael Vogel wrote:
Jim Van Meggelen schrieb:
YIKES! What kind of processor have you got there?
Its a:
- Pentium II (Deschutes) 333MHz
- 128mb memory
I'm using it as:
- Mailserver (IMAP, SMTP)
- Webserver (mainly for webmail)
- Newsserver
-
On Wednesday 15 December 2004 23:24, Puddle wrote:
Thanks, that makes a lot more sense. Would VoIP
phones still require FXO units or would that not
require any special telephony hardware?
SIP phones connect by ethernet - no telephony hardware needed.
You would want an FXO port if you want
Hi.
Does anyone know of any small SIP phones (and preferably have some experience
of using them and happy to recommend them)?
By 'small' I mean a single-piece phone, with dial buttons in the handset, so
that it can be carried around easily in a laptop bag. Something like
On Sunday 12 December 2004 20:12, Clay Reiche wrote:
I don't know of a small phone, but you use a WorlACCXX TA200 device (pretty
small) along with any standard analogue phone.
http://www.worldaccxx.com I have one and carry it around in my laptop bag.
Demensions are 6x4.5x1.25
Thanks. In
[mailto:[EMAIL PROTECTED]
On Sun, 2004-12-12 at 21:27, Antony Stone wrote:
Thanks. In fact I already have a Grandstream ATA-486, which I'm
very pleased with: http://www.grandstream.com/y-ht486.htm This unit is
even smaller - 105 x 75 x 25mm (or 4 x 2.75 x 1), however I'm just
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