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 mypa
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?
menuselect
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 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 Aste
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.
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 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 Switc
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
> sampl
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 greeting,
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 netwo
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 networks
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.
>
> I'm
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 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 run
On Monday 26 June 2017 at 18:01:22, J Montoya or A J Stiles wrote:
> On Monday 26 Jun 2017, 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 le
Hi.
I have some Nagios / Icinga monitoring plugins I've created for Asterisk, and
one of them checks the percentage of SIP accounts which are currently
registered on an Asterisk server.
It does this by running "sip show peers" via AMI and analysing the summary
line at the end:
1066 sip peers
On Friday 30 June 2017 at 19:11:08, Jonathan H wrote:
> I use a python AGI which pulls some info from a web service, which should
> take half a second.
>
> Sometimes, it takes 5-10 seconds which blocks the dialplan execution, but
> the dialplan should continue immediately as it's not dependent on
Hi.
I'm trying to get a list of the channels currently in use on an Asterisk server
(1.8.32.1 if it matters) over AMI.
I send the command "sip show channels", and I get back a response along the
lines of (* used to protect the innocent...):
Peer User/ANR Call ID Fo
u will probably achieve your goals
>
> https://wiki.asterisk.org/wiki/display/AST/AMI+Actions
Hm, I don't see anything there which will give me a list of the SIP channels
currently in use - what command should I be using for that?
Thanks,
Antony.
> On 7 Jul 2017 10:32 pm, Antony Stone wr
On Saturday 08 July 2017 at 10:16:19, Antony Stone wrote:
> On Saturday 08 July 2017 at 07:15:08, Marcelo Terres wrote:
> > There are no sip show channels on AMI. Also, the output that you sent is
> > not a AMI output. Are u using AMI ou running commands on console?
>
>
On Friday 14 July 2017 at 23:34:37, Motty Cruz wrote:
> Since the upgrade our remote users' conversions are choppy.
> Monitoring using CLI, I noticed the device always select ulaw
> for codec.
What's the device?
What are its codec settings?
What's your available & used bandwidth on the server'
On Thursday 20 July 2017 at 20:46:30, Marcelo Terres wrote:
> I don't have much knowledge about freepbx, but if some day I had to use it,
> I would prefer to use the Asterisk compiled from source, unless it comes
> with an Asterisk package (rpm, supposing it is running CentOS).
FreePBX (as a dis
On Saturday 29 July 2017 at 19:03:55, Joshua Colp wrote:
> On Sat, Jul 29, 2017, at 02:55 PM, O. Hartmann wrote:
> > Scenario:
> >
> > Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to
> > 192.168.254.1:5060) is behind
> > a NAT, acting as a client to our ITSPs SIP server. But also, this
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
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
http://maplin.co.uk/
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 6"x4.5"x1.25"
Thanks.
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
> allo
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 (whet
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 t
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 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 detec
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 inform
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 T
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.
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 ha
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 o
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 s
From: Florian Overkamp [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
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)
> - Newsserv
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
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.
>
>
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
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 peop
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
u
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 developm
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 SI
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 bandwidt
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
On Sunday 19 December 2004 10:13, hadi wrote:
> Yes
> U can do it with asterisk and by dialing *98 on your Ip Phone you can
> listen to your message
No, that's voicemail. ie: The caller leaves a message and hangs up, then you
retreive the message later.
The OP wanted to be able to hear the inc
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.
> >
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,
> >>> ge
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 Prod
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 p
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