Re: [asterisk-users] How to know the IP of "manager show connected" in dialplan

2018-07-25 Thread David Duffett
How about using the CUT() function to get the IP address from the return
from running the System() application running asterisk -rx "manager show
connected"?

I'm not in front of a machine, so cannot test this out...



On Wed, 25 Jul 2018, 15:42 Ludovic Gasc,  wrote:

> Maybe I'm wrong, but, with the information you give us, for me, it seems
> more elegant to use FastAGI to be sure to communicate with the right remote
> process.
>
> As Antony suggested, UserEvent is also an option, except if you have
> several dialers connected at the same time or if you need to have an
> acknowledge that the action is correctly launched.
>
> Regards.
> --
> Ludovic Gasc (GMLudo)
>
>
> Le mer. 25 juil. 2018 à 19:54, Saint Michael  a écrit :
>
>> ​I need to launch a remote process at the machine that has the dialer. I
>> could
>> hard-code the IP address in a global variable, but It would be much more
>> elegant if the dialplan would have a "manager" object where I could read
>> "manager-->connected". ​
>>
>>
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Re: [asterisk-users] Blocking outgping caller id on a PRI E1

2017-11-08 Thread David Duffett
It is likely being set by your PRI provider.
Contact them to investigate.

All the best...

On 8 Nov 2017 9:03 am, "Neil Youngman"  wrote:

> I am trying to block/hide outgoing caller id on a PRI E1.
>
> It seems that it should be fairly simple, but it is defeating me.
>
> https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says:
> "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)"
>
> That doesn't seem to do it.
>
> The calls are originated from AMI and I have tried a blank "CallerId:"
> line and "CallerId: <>"". Neither of those made any difference.
>
> I have also tried "hidecallerid=yes" in chan_dahdi.conf, but that has also
> made no difference.
>
> I assume that I am missing something obvious?
>
> Neil Youngman
>
>
> Neil Youngman
> Developer
> Wirefast Limited
>
> Wirefast provides secure corporate messaging services.
> See our messaging solutions at  http://www.wirefast.com/
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Re: [asterisk-users] TDM400P takes too long to ring

2017-04-27 Thread David Duffett
If you are trying to detect caller ID, and it is being supplied by the
telco in the format you have configured in /etc/chan_dahdi.conf then this
should not cause a delay. Are you actually seeing the caller ID being
displayed on the ringing phones?

If, however, the telco is not supplying caller ID info, or it is being
supplied in a format that you have not set up for, this is likely the cause
of the delay (looking for caller ID).

All the best,

David

On 27 April 2017 at 12:48, Ryan, Travis <ry...@oscarwinski.com> wrote:

> Hey all,
>
>
>
> I have a setup with two analog lines coming into and Asterisk 13 box with
> a TDM400P and it takes a lot of rings before asterisk takes over. I’ve
> traced this same box on two different analog providers so it probably isn’t
> a problem with them.
>
>
>
> I DO have callerid enabled and not sure I can turn it off (if they will
> let me). Any other ideas of making it ring through faster?
>
>
>
> By the time my internal phones get rang the customer has heard upwards of
> 7 rings. Some customers hang up thinking no one will answer.
>
>
>
> Also, I have fax detection off in my dialplan.
>
>
>
> Thanks,
>
> Travis
>
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Re: [asterisk-users] Incoming Call by DID

2016-10-26 Thread David Duffett
It seems like your SIP provider is not sending and DID information, or that
the information is not being sent in the same format you are using in your
dialplan.

You can check this by looking at the SIP debug information for the inbound
calls and/or by checking with your SIP provider (that they are sending the
DID number and what format it is in).

All the best,
David

On 27 Oct 2016 5:21 am, "KyD"  wrote:

Hi,

My sip provider gave me 2 numbers for the incoming call via pstn.

nro1 = 12341234
nro2 = 45674567

I have a dialplan for each.
if i put this on my dialplan:

exten => s,1,Dial(SIP/1001)
exten => Hangup()

Works!

But if i put one of them:

exten => 12341234,1,Dial(SIP/1001)
exten => _1234,1,Dial(SIP/1001)
exten => 45674567,1,Dial(SIP/1001)
exten => _4567,1,Dial(SIP/1001)

incoming calls do not arrive.

Any ideas?
--
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Quanto mais você sabe, mais você percebe que você não sabe nada.

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Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-24 Thread David Duffett
Yes, sorry, my idea was too simplistic, as it did not take into account
that the caller would already be hearing the ringing and therefore the
announcement would need to be somehow mixed with that ringing...

On 24 August 2016 at 08:27, Jean Aunis <jean.au...@prescom.fr> wrote:

> Using Progress didn't solve the problem. If I cannot find another way, I
> will use your solution of recording the ring tone.
>
> Le 23/08/2016 à 20:53, Israel Gottlieb a écrit :
>
> Maybe try progress() instead of answer ()
>
> בתאריך 23 באוג׳ 2016 7:19 PM,‏ "Jean Aunis" <jean.au...@prescom.fr> כתב:
>
>> Thank you, I just tried your suggestion. Strangely, the announcement is
>> played only if I try to dial a SIP peer which is not available (not
>> registered to be more precise). If the SIP peer is available, I only get
>> the ring tone, and never hear the announcement. Here is the dialplan (I had
>> to add an Answer() before the Dial, otherwise the announcement is never
>> played, even in the first case) :
>>
>> exten = 007,1,Answer()
>> same  = n,Dial(SIP/foo/s@playme,40)
>>
>> [playme]
>> exten = s,1,Ringing()
>> same  = n,Wait(10)
>> same  = n,Playback(/var/lib/asterisk/sounddir/announce,noanswer)
>> When it is working, I can see the following output in the CLI, which is
>> not there otherwise :
>> -- SIP/x requested media update control 26, passing it to
>> Local/s@playme-05be;1
>>
>> Otherwise, no error message, Asterisk tells he is playing the
>> announcement but I don't hear it.
>>
>> Best regards
>>
>> Jean Aunis
>>
>
>
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Re: [asterisk-users] Dial and start music on hold after timeout

2016-08-23 Thread David Duffett
tch
>> manure, program a computer, cook a tasty meal, fight efficiently, die
>> gallantly. Specialization is for insects.
>> ---Heinlein
>>
>>
>>
>>
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[asterisk-users] Don't Miss Out - AstriCon is NEXT MONTH!

2016-08-22 Thread David Duffett
If you, like me, have a tendency to leave things until the last minute, you
may be in danger of missing out on AstriCon this year - as it is a little
earlier than usual, September 27-29 in Glendale, AZ. www.astricon.net

Three days of nothing but Asterisk-themed talks, demos and Expo exhibits
are bound to help you learn more, progress your projects, perfect your
solutions or even find customers for your new Asterisk-related thing!

Key member of the Asterisk development team will be on hand to chat and
answer questions, as well as presenting (along with Matt Jordan, Digium
CTO) some of what they have been up to in our closing keynote session...

Talking of keynotes, we have two more top class addresses this year, from
Bill Ledingham, CTO/EVP of Engineering at Black Duck Software and Chris
Matthiue, Director of IoT Engineering at Citrix. BTW, these will not be
officially announced until later today, but I wanted you to know first!
You really don't want to miss those... www.astricon.net

In addition to all of the above, there are more projects than ever before
represented in our Open Source Showcase this year - including Kamailio,
FreeSWITCH, Homer, CGRates and OpenSIPS!
As always, Asterisk initial creator, Mark Spencer and Allison Smith, the
first lady of Asterisk will be on hand to participate in all the fun.

As well as coming to AstriCon yourself, you could always share the love by
bringing some colleagues with you, using the excellent Diamond Pass. Full
details at www.astricon.net

Hope I get to see you in Glendale!

All the best,

David
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Re: [asterisk-users] Help With Physical Layer

2015-06-30 Thread David Duffett
What response do you get to *CLI pri show spans ?

On 30 June 2015 at 09:34, Tony Kasule timotsm...@gmail.com wrote:

 Hello,

 Anyone to help me with this issue? It has never worked :(

 On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com
 wrote:

 Hello users,

 I have a Digium Te235 and asterisk 13  which have worked well with 1
 carrier but we have failed to add a 2nd carrier. The second telco brings
 their E1 line over finer, terminated in a RAD modem and they give me
 ethernet to the E1 card. It's the first time i am having install such a
 solution, which ideally would be not a big problem.

 However, The  physical layer has failed to come up! I have tried the same
 setup in an Alcatel OmniOCX and it works well. I can confirm that the port
 is also well configured because when I interchange the cables (with the
 exiting cable from the other telco), the alarm clears emmediately for the
 1st telco and becomes RED for the 2nd telco. A loop also clears the alarm
 on both ports.

 The telco has told me to make sure that Line Impedance is 120 ohms but
 There's no where to set that and when I was reading, I was told that E1 is
 already 120 ohms so no need to change anything.

 Has anyone here experienced this?  What other things can I try?

 Thank you!

 Regards,
 Tim



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Re: [asterisk-users] Missed call

2015-06-05 Thread David Duffett
On some SIP phones it is possible to turn off the missed call
notifications, but I am not aware of a way to do the same on any cell
phones.
On 5 Jun 2015 07:29, Mehmet Avcioglu meh...@activecom.net wrote:


 There is no signal that is sent to display a missed call. Your cell phone
 does that. If it rings and is not answered it counts that as a miss. The
 only way to avoid it is to not ring it. So instead of simultaneous ringing
 you can do sequential.

 --
 Mehmet Avcioglu
 meh...@activecom.net

  On Jun 4, 2015, at 11:21 PM, Luca Bertoncello lucab...@lucabert.de
 wrote:
 
  Hi list!
 
  I configured Asterisk to forward the incoming call for a number to both
 phones.
  I wrote that:
 
  exten = _0049351222,n,Dial(SIP/0049351222SIP/004935,,R)
 
  of course it works...
  Now the problem is, that when a phone get the call, on the other phone I
 get 1 missed call...
  Is it possible to avoid that and signaling the other phone, that the
 call was not missed?
 
  Thanks a lot
  Luca Bertoncello
  (lucab...@lucabert.de)
 
 
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Re: [asterisk-users] PRI Callerid Passthrough

2015-03-18 Thread David Duffett
In a word, no.

PRI service providers will generally only allow the caller ID to be set to
one of the numbers in the range that you have for inbound with them.
On 18 Mar 2015 11:30, Rizwan H Qureshi rizwanhas...@gmail.com wrote:

 Hi All,
 I have to forward incoming call on PRI back out to PRI but I need the
 original Callerid to passthrough. Is it possible with DAHDI PRI cards
 without involving the service provider?

 Thanks

 --
 Best Ragards
 Rizwan H Qureshi

 V: +971 (0) 528272154
 linkedin.com/in/rhqureshi



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Re: [asterisk-users] Understanding the right way to get started with multiple trunks/extensions

2015-03-04 Thread David Duffett
If you would like to set things up via the GUI on your incredible PBX, you
could use queues instead of call groups (making your SIP clients agents of
the appropriate queues), and in the queues configuration page there is an CID
Name Prefix option, which allows you to add a label that will show up as
part of the caller ID - so you will see it as the call comes in.

If, on the other hand, you want to achieve your aim through native
configuration files, you could add a line like:
exten = *home-number*,1,Set(CALLERID(name)=Home)
exten = *home-number*,n,*continue handling call as you were before*

This way, you will be setting the caller ID with a name label that can be
observed on the SIP client before answering.

All the best,

David



On 4 March 2015 at 11:53, Mark Rogers m...@more-solutions.co.uk wrote:

 Background: I dabbled with asterisk years ago, and more recently have
 more-or-less functioning IncrediblePBX systems for experimenting, but
 I want to understand more so I'm now working with distro packages
 (Ubuntu) and hand edited configurations files.

 I have three SIP trunks, each providing me with a UK telephone
 number. They are for my home (1), my wife's home-based bookkeeping
 business (2), and for taking support calls on from work (3). For
 handsets I only really have mobile phones running SIP/IAX clients,
 although I might put a real desktop handset on my wife's desk at
 some point.

 The objective is that my wife picks up calls to (1) or (2), I pick up
 calls to (2) or (3), and maybe a colleague might also pick up calls to
 (3).

 I can see how to do this with call groups, but I would like to be able
 to see at the handset which trunk the call has come in on, which I
 don't think I can do that way?

 Or I could just set up multiple connections in the SIP/IAX client, so
 each trunk has its own extension and we each connect to multiple
 extensions as required. That has the added advantage of making the
 voicemail relate to which trunk the call comes in on, which is also
 more appropriate here.

 However, the latter option feels wrong somehow and I haven't found
 any examples suggesting it's the right way to do things.

 Also, I am keen to minimise battery overhead on the mobile phones - is
 linking to multiple accounts is less efficient?

 PS: Although I only really care about inbound calls at the moment, I
 daresay I will want to handle internal calls at some point too,
 although nothing stops me having dedicated per-user extensions as well
 as per incoming trunk.

 Mark
 --
 Mark Rogers // More Solutions Ltd (Peterborough Office) // 0844 251 1450
 Registered in England (0456 0902) 21 Drakes Mews, Milton Keynes, MK8 0ER

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Re: [asterisk-users] Astricom 2014 presentations

2014-10-29 Thread David Duffett
I can confirm that all the videos from AstriCon 2014 will be available at
www.AstriCon.net within about 3 weeks.
On 29 Oct 2014 16:33, Jeff LaCoursiere j...@jeff.net wrote:


 On 10/29/2014 05:50 AM, Bogdan Cristea wrote:

 Hi

 Will the presentations made at Astricom 2014 be made public as recorded
 videos ?

 thanks
 Bogdan


 I'll second the request for that, and also ask if the sessions on Kamailio
 will be similarly available.

 Cheers,

 j

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Re: [asterisk-users] allo.com gsm card with AsteriskNOW

2014-10-15 Thread David Duffett
On 15 October 2014 11:59, Nicolas Pham Van Huyen nico...@phamvanhuyen.com
wrote:

 Dear all,

 I'm searching someone who already installed allo.com gsm card with
 Asterisk NOW.

 I installed the hardware in my new server, but when runing dahdi_genconf I
 always have the no span message.
 I tried to install the driver according to allo.com doc, but it seems not
 up to date, and maybe done more specificly for an appliance..

 [root@localhost ~]# dahdi_genconf
 Empty configuration -- no spans
 Empty configuration -- no spans
 Empty configuration -- no spans

 Thanks for help
 Nico

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Step 1 would be an 'lspci' on the Linux command line to see if the Linux
box recognises the card
Step 2 would be to ensure that your DAHDI version is new enough to work
with the card

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6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
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Re: [asterisk-users] OT: Question on Caller ID (Spoofing calls with Asterisk)

2014-08-26 Thread David Duffett
Asterisk can set any Caller ID name and number you want with the CALLERID()
function.

Type 'core show function CALLERID()' on the Asterisk command line to get
the details.

Whether your trunk will convey the Caller ID you have set is another matter
- and entirely the choice of your outbound carrier.


On 26 August 2014 22:45, Jeffrey Walton noloa...@gmail.com wrote:

 I got a call from an overseas call center telling me about the
 problems with the Windows machine I was using. They wanted to remote
 in and fix things for me ... (Ignore the fact I use a MacBook Pro or
 an ASUS laptop with Debian).

 What I found curious was the caller's name was Asterisk, and the
 caller's number was Asterisk@10 or or Astrk@10 similar. (I don't
 recall the exact number, but it was malformed and it had an '@' in
 it).

 I'd like to read a little more about spoofing calls with Asterisk. Can
 anyone provide a reference?

 Thanks in advance.

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Re: [asterisk-users] Connecting Asterisk and BT Versatility PBX via NT BRI port

2014-08-20 Thread David Duffett
Great news!

On Wednesday, August 20, 2014, Roberto Fichera ker...@tekno-soft.it wrote:

  On 08/04/2014 03:03 PM, David Duffett wrote:

 Please come back to let us know if this actually does fix the issue.


 So far so good the external voltage supply for the OpenVOX card has
 arrived and I can confirm
 that the BT Versatility PBX worked like a charm. DAHDI channel has been
 configured in PtP mode.

 Cheers,
 Roberto Fichera.

  On 4 Aug 2014 14:36, Roberto Fichera ker...@tekno-soft.it
 javascript:_e(%7B%7D,'cvml','ker...@tekno-soft.it'); wrote:

  On 08/04/2014 01:21 PM, David Duffett wrote:

 If the OpenVox card can supply the voltage, then it will a configuration
 option (probably in hardware, like some jumpers) of the card itself.


 Yep! The OpenVOX card has a jumper for each port and a connector where to
 insert the external voltage supply device.

  I was going to point you to the Xorcom Astribank, which I know supplies
 the required voltage.


 Ah! Ok! I was thinking you was giving me something like a temporary
 solution for supply the voltage to the PBX.

 Cheers,
 Roberto Fichera.

  All the best,

 David
 On 4 Aug 2014 13:16, Roberto Fichera ker...@tekno-soft.it
 javascript:_e(%7B%7D,'cvml','ker...@tekno-soft.it'); wrote:

  On 08/03/2014 04:37 PM, David Duffett wrote:

 Does the BT Versatility system you are trying to connect require a
 voltage on the line, which it would normally get from the BT connection?

  Some BRI equipment does, and some does not, and it may be the
 Panasonic system you refer to is happy without the voltage, while, perhaps,
 the BT Versatility needs it.

  If you find out that it does need the voltage, I do not think the
 OpenVox card you mentioned will supply it, but I can point you to a
 solution if required.


 Just checked if the BT Versatility takes or not the voltage from the
 NT-1 and it does! So the problem looks really this!

 Can you point me to how supply voltage to the PBX while I'm waiting the
 given OpenVox adapter being delivered?

 Cheers,
 Roberto Fichera.


  All the best,



 On Sunday, August 3, 2014, Mc GRATH Ricardo mcgra...@mail2web.com
 javascript:_e(%7B%7D,'cvml','mcgra...@mail2web.com'); wrote:

 Hi, it seems have problem with physical issue layer 1, first check
 wiring connection, by the way  you can check with card led (If ISDN  plugs
 into the port, the LED will not blink, but in red color.) after  dahdi
 alarm status, and dahdi restart command.
 At last should be signalling mode to signalling = bri_cpe_ptmp
 Good luck

 Mc GRATH Ricardo
 E-Mail mcgra...@mail2web.com
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 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX
  · UK
 direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
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Re: [asterisk-users] The plain old PBX functionality

2014-08-08 Thread David Duffett
Using the BLFs on Digium phones does not require the use of the Digium
Phone Module for Asterisk, or DPMA. SchmoozeCom (the FreePBX guys) use the
BLFs on Digium phones independently of the DPMA.

I am not sure why a previous response refers to this module as 'toxic'. It
is a free to use module which allows a host of Digium phone features to be
quickly implemented with Asterisk, like security-enhanced auto provisioning.

Digium phones are the only SIP phones in the world designed specifically to
work with Asterisk, and each purchase helps to fund the ongoing development
of open source Asterisk. In addition, they have a JavaScript API that
allows anyone that wants to create custom apps on the phone to do so.

They can, of course, work with any standard SIP system and can be
provisioned using FTP/HTTP(S) and XML files, just like any other phone.

On Friday, August 8, 2014, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 On Friday 08 Aug 2014, Gergo Csibra wrote:
  Hi,
 
  back in the old analog telephony days there was digital PBX-es and
  digital system phonesets. This phonesets have had many individual
  illuminatable buttons connected with extensions. The PBX can show on
  the buttons if some extension is ringing (blinks) or busy (constant
  light), and the user can transfer the call with one touch (pressing
  one of this button).
 
  I search this functionality in Asterisk. What versions, and what
  extension functions (or other settings), and what VoIP phones can do
  this?

 What you are looking for is a Busy Lamp Field  (BLF).  This can usually
 be
 controlled by custom SIP headers  (though naturally, every manufacturer's
 implementation is ever so slightly different).

 N.B.  Be sure to *avoid* Digium phones if you want BLF functionality!
  Making
 use of this feature would require you to pollute your system by installing
 a
 toxic, binary-only module.

 --
 AJS

 Note:  Originating address only accepts e-mail from list!  If replying off-
 list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] Connecting Asterisk and BT Versatility PBX via NT BRI port

2014-08-04 Thread David Duffett
If the OpenVox card can supply the voltage, then it will a configuration
option (probably in hardware, like some jumpers) of the card itself.

I was going to point you to the Xorcom Astribank, which I know supplies the
required voltage.

All the best,

David
On 4 Aug 2014 13:16, Roberto Fichera ker...@tekno-soft.it wrote:

  On 08/03/2014 04:37 PM, David Duffett wrote:

 Does the BT Versatility system you are trying to connect require a voltage
 on the line, which it would normally get from the BT connection?

  Some BRI equipment does, and some does not, and it may be the Panasonic
 system you refer to is happy without the voltage, while, perhaps, the BT
 Versatility needs it.

  If you find out that it does need the voltage, I do not think the
 OpenVox card you mentioned will supply it, but I can point you to a
 solution if required.


 Just checked if the BT Versatility takes or not the voltage from the NT-1
 and it does! So the problem looks really this!

 Can you point me to how supply voltage to the PBX while I'm waiting the
 given OpenVox adapter being delivered?

 Cheers,
 Roberto Fichera.


  All the best,



 On Sunday, August 3, 2014, Mc GRATH Ricardo mcgra...@mail2web.com wrote:

 Hi, it seems have problem with physical issue layer 1, first check wiring
 connection, by the way  you can check with card led (If ISDN  plugs into
 the port, the LED will not blink, but in red color.) after  dahdi alarm
 status, and dahdi restart command.
 At last should be signalling mode to signalling = bri_cpe_ptmp
 Good luck

 Mc GRATH Ricardo
 E-Mail mcgra...@mail2web.com
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 Digium, Inc. · Director, Worldwide Asterisk Community
 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX ·
 UK
 direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
 Check us out at: http://digium.com · http://asterisk.org
 http://www.asterisk.org/





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Re: [asterisk-users] Connecting Asterisk and BT Versatility PBX via NT BRI port

2014-08-04 Thread David Duffett
Please come back to let us know if this actually does fix the issue.
On 4 Aug 2014 14:36, Roberto Fichera ker...@tekno-soft.it wrote:

  On 08/04/2014 01:21 PM, David Duffett wrote:

 If the OpenVox card can supply the voltage, then it will a configuration
 option (probably in hardware, like some jumpers) of the card itself.


 Yep! The OpenVOX card has a jumper for each port and a connector where to
 insert the external voltage supply device.

  I was going to point you to the Xorcom Astribank, which I know supplies
 the required voltage.


 Ah! Ok! I was thinking you was giving me something like a temporary
 solution for supply the voltage to the PBX.

 Cheers,
 Roberto Fichera.

  All the best,

 David
 On 4 Aug 2014 13:16, Roberto Fichera ker...@tekno-soft.it wrote:

  On 08/03/2014 04:37 PM, David Duffett wrote:

 Does the BT Versatility system you are trying to connect require a
 voltage on the line, which it would normally get from the BT connection?

  Some BRI equipment does, and some does not, and it may be the Panasonic
 system you refer to is happy without the voltage, while, perhaps, the BT
 Versatility needs it.

  If you find out that it does need the voltage, I do not think the
 OpenVox card you mentioned will supply it, but I can point you to a
 solution if required.


 Just checked if the BT Versatility takes or not the voltage from the NT-1
 and it does! So the problem looks really this!

 Can you point me to how supply voltage to the PBX while I'm waiting the
 given OpenVox adapter being delivered?

 Cheers,
 Roberto Fichera.


  All the best,



 On Sunday, August 3, 2014, Mc GRATH Ricardo mcgra...@mail2web.com
 wrote:

 Hi, it seems have problem with physical issue layer 1, first check
 wiring connection, by the way  you can check with card led (If ISDN  plugs
 into the port, the LED will not blink, but in red color.) after  dahdi
 alarm status, and dahdi restart command.
 At last should be signalling mode to signalling = bri_cpe_ptmp
 Good luck

 Mc GRATH Ricardo
 E-Mail mcgra...@mail2web.com
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 To UNSUBSCRIBE or update options visit:
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 [image: Digium logo]
 *David Duffett*
 Digium, Inc. · Director, Worldwide Asterisk Community
 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX
  · UK
 direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
 Check us out at: http://digium.com · http://asterisk.org
 http://www.asterisk.org/





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Re: [asterisk-users] Connecting Asterisk and BT Versatility PBX via NT BRI port

2014-08-03 Thread David Duffett
Does the BT Versatility system you are trying to connect require a voltage
on the line, which it would normally get from the BT connection?

Some BRI equipment does, and some does not, and it may be the Panasonic
system you refer to is happy without the voltage, while, perhaps, the BT
Versatility needs it.

If you find out that it does need the voltage, I do not think the OpenVox
card you mentioned will supply it, but I can point you to a solution if
required.

All the best,



On Sunday, August 3, 2014, Mc GRATH Ricardo mcgra...@mail2web.com wrote:

 Hi, it seems have problem with physical issue layer 1, first check wiring
 connection, by the way  you can check with card led (If ISDN  plugs into
 the port, the LED will not blink, but in red color.) after  dahdi alarm
 status, and dahdi restart command.
 At last should be signalling mode to signalling = bri_cpe_ptmp
 Good luck

 Mc GRATH Ricardo
 E-Mail mcgra...@mail2web.com javascript:;
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Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
Check us out at: http://digium.com · http://asterisk.org
http://www.asterisk.org/
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Re: [asterisk-users] Asterisk on Windows

2013-12-04 Thread David Duffett
I just checked my calendar, and - surprisingly - it's not April 1st!
On 4 Dec 2013 23:55, Gregory Malsack gmals...@coastalacq.com wrote:

 I second that!

 *Sent from my Verizon Wireless 4G LTE DROID*


 Eric Wieling ewiel...@nyigc.com wrote:

 Asterisk is Open Source, any company can port Asterisk to Windows.
 Nobody has.  Personally, I don't want Digium taking valuable and limited
 development resources to create a Windows port.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of CDR
 Sent: Wednesday, December 04, 2013 10:19 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Asterisk on Windows

 Digium is 100% lost in the map. If they would come up with a Paid version
 of Asterisk, one that would use the .NET framework in Windows, something
 simple to install, they could go public on the product.
 Linux has a very steep learning curve. A Windows application that would do
 exactly the same would be a home run. Note: I am a Linux expert user, but
 it took me years to get here. And still, moving from regular RHEL 6.0 to
 Fedora 20 (RHEL 7) is a pain in the neck. The .NET framework and Windows
 server 2012 are miles away in terms of friendliness and on equal footing on
 performance. I don´t mean another slow cygwin port, I man a native Asterisk
 for windows. In fact, I would invest on the project if somebody wants to do
 it.

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Re: [asterisk-users] sipgate outgoing calls

2013-09-18 Thread David Duffett
I believe registration is in place, otherwise inbound calls would not work.

Also, registration is not required for outbound calls to work.

I would suggest cutting down your sip.conf profile to this minimal
configuration:

host=sipgate.co.uk
username=xxx
fromuser=xxx
insecure=invite,port
secret=xxx
context=my-inbound-context
type=peer

If outbound calls still do not with this, I would suggest that there may be
an issue in the general section of your sip.conf

Assuming calls do work, you can then add any other configuration lines you
feel are necessary - but remember, as with all Asterisk configuration
files, less is more :-)
 On 18 Sep 2013 22:06, Administrator TOOTAI ad...@tootai.net wrote:

 Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit :

 Hello


 Hi


 i am trying to setup sipgate gateway

 i can get incoming calls fine, but when i dial in and then try to dial
 out i get this in asterisk command line

 -- Called 01179248615@sipgate
 [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885
 handle_response_invite: Failed to authenticate on INVITE to
 '01179553708 sip:sip...@sipgate.co.uk;**tag=as30eb9dd1'
 -- SIP/sipgate-014d is circuit-busy
   == Everyone is busy/congested at this time (1:0/1/0)


 here is my sip.conf file


 [general]
 port = 5060
 bindaddr = 0.0.0.0
 context=default
 qualify=no
 disallow=all
 allow=alaw
 allow=ulaw
 allow=g729
 allow=gsm
 allow=slinear
 srvlookup=yes
 videosupport=yes
 alwaysauthreject=yes

 register = 
 SIP-ID:SIP-Password@sipgate.**co.uk/SIP-IDhttp://SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID

 [sipgate]
 type=peer
 secret=SIP_PASSWORD
 insecure=invite
 username=SIP-ID
 defaultuser=SIP-ID
 fromuser=SIP-ID
 context=sipgate_in
 fromdomain=sipgate.co.uk
 host=sipgate.co.uk
 outboundproxy=proxy.live.**sipgate.co.ukhttp://proxy.live.sipgate.co.uk
 qualify=yes
 disallow=all
 allow=alaw
 dtmfmode=rfc2833


 SIP-ID:SIP-Password
 obviously, i replace these with my login details

 but, are these the same thing ?
 SIP-Password
 SIP_PASSWORD

 the sipgate guides are contradictory

 http://www.sipgate.com/faq/**article/394/How_do_I_**configure_Asteriskhttp://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
 http://www.live.sipgate.co.uk/**faq/article/508/How_do_I_**
 configure_Asterihttp://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri
 sk


 any suggestions ?

 Many thanks


 My setup with sipgate.de

 [sipgate]
 type=peer
 secret=MY-PASSWORD
 defaultuser=SIP-ID
 host=217.10.79.9
 fromuser=SIP-ID
 fromdomain=sipgate.de
 context=incoming-sipgate
 ;qualify=900
 dtmfmode=info
 directmedia=yes
 insecure=port,invite
 disallow=all
 allow=ulaw,alaw
 accountcode=MY-ACCOUNTCODE

 What you forget is to register with them:

 ; Sipgate
 register = 
 SIP-ID:my-passw...@sipgate.de/**SIP-IDhttp://SIP-ID:my-passw...@sipgate.de/SIP-ID;don't
  accept to register without FQDN

 Hope that help

 --
 Daniel

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Re: [asterisk-users] Dedicated hangup extension h

2013-08-28 Thread David Duffett
I would set a no-use flag in all extensions that you do not want to use
the h, and then test for it in the h extension itself - if it is set you
could just run the Hangup application.
On 28 Aug 2013 08:51, Grant Bagdasarian g...@cm.nl wrote:

 Hello,

 ** **

 We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming
 calls from our carrier.

 ** **

 The sip.conf looks like this:

 ** **

 [kamailio1]

 type=friend

 host=10.0.0.1

 context=incoming

 disallow=all

 allow=alaw

 ** **

 All calls hit the incoming extension. In the extensions.conf we have
 multiple extensions configured, but now I have to add one which uses the
 special h extension to perform a CURL action whenever the user hangs up.
 The problem is that once I’ve registered a h extension, it is executed for
 all extensions in the incoming context.

 ** **

 exten = _X.,1,Playback(invalid)

 exten = _X.,n,Hangup

 ** **

 exten = 1000,1,Playback(welcome)

 exten = 1000,n,Read(dtmfinput,15)

 exten = 1000,n,Hangup

 ** **

 exten = h,1,Set(response=${CURL(
 http://sample.company.local/PostHandler.ashx,var1=${dtmfinput}var2=1000)}
 )

 ** **

 Is it possible to give each extension its own h extension? If not, is
 there another way to do this?

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Re: [asterisk-users] Asterisk stops registering

2013-07-03 Thread David Duffett
I would enable SIP debugging, but only for that provider.

This can be done on the Asterisk command line by using either of the
following:

* sip set debug peer Your VoIP provider peer name

or

* sip set debug ip the ip address of your VoIP provider


On 3 July 2013 23:25, Ian Pilcher arequip...@gmail.com wrote:

 On several occassions lately, my home Asterisk box has stopped
 registering with my VoIP provider.  I haven't been able to reproduce the
 problem, and the log doesn't contain anything useful.

 How can I increase the log verbosity for SIP registration-related
 events?  I've looked through logger.conf and tried searching with
 Google, but I haven't been able to find a clear answer.

 This is Asterisk 1.8.20.0, BTW.

 Thanks!

 --
 
 Ian Pilcher arequip...@gmail.com
 Sometimes there's nothing left to do but crash and burn...or die trying.
 


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*David Duffett*
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6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
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[asterisk-users] AstriCon 2013 (our 10th AstriCon) needs YOU!

2013-05-16 Thread David Duffett
As we plan for our 10th AstriCon, which will be in Atlanta, GA the week 
commencing October 7th, we want to make sure that our conference sessions are 
the best they've ever been!


That's why we need YOU to submit a speaking proposal - to share you experiences 
and ideas around Asterisk!


The best sessions at AstriCon are always those from the Community that show 
what you've been up to with Asterisk, new ways that you've put Asterisk to 
work, etc. - where you're showing a use case or some additional stuff that 
you've developed within, or in addition to, Asterisk - we're waiting to hear 
from you...


If your speaking submission is accepted, you will get a free all-access pass 
pass for the whole event - worth $495 (and that's the early bird price, lasts 
until the end of June).


Please make your speaking submission at 
https://docs.google.com/forms/d/1Li-hRCtqOQHpl6RHMt0PX6SvWGbwqPfKsOfRm35Vkz8/viewform


If you are a developer, do remember our AstriDevCon get-together which is 
traditionally on the Monday - this is where the Community discusses new 
features and the priorities are set for future Asterisk releases.


At AstriCon, we also seek to recognise those in the Community that have made 
great contributions - if you have any nominations for this year, please let me 
know.


REMEMBER: If you're serious about Asterisk, you'll be at AstriCon! Registration 
is at http://www.asterisk.org/community/astricon-user-conference/register


All the best,


David


Digium logo
David Duffett
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
Check us out at: http://digium.com · http://asterisk.org
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[asterisk-users] Speaking opportunities at Digium Asterisk World/IT Expo - Miami Beach - 1/31 and 2/1

2013-01-02 Thread David Duffett

Happy New Year.

Digium Asterisk World is a set of conference sessions that run at IT Expo in 
Miami Beach, FL at the end of this month.

See http://www.tmcnet.com/voip/conference/digium-asterisk-world/default.htm  
for more details.

The conference runs 1/30-2/1, and we have a small number (2 or 3 ) of speaking 
slot available, over 1/31 and 2/1.

This presents an opportunity for you to talk about your contribution(s) to the 
Asterisk project or describe your interesting Asterisk implementations - 
identifying you as a thought-leader in your field.

Digium Asterisk World is more commercially focussed than AstriCon, and a 
proportion of the audience will be looking for solutions, rather than 
specifically wanting to get directly involved in the implementation itself - so 
this may provide a way for you to meet customers, or to show the community what 
you have been up to with Asterisk.

The sessions must be educational and non-commercial. No pitches.  

Each session is 45 mins in duration, including time for Q and A.

As a thank you for your successful speaking submission, we will give you an 
all-access pass to IT Expo, which includes all conference sessions (for all 
tracks at IT Expo) and conference meals.

[You will need to make your own arrangements to get to the conference and for 
local accommodation, if needed - we will not fund or contribute to this].  

Please get in touch with me directly if you would like to speak - sending a 
session title and description and a few lines about you, as a speaker.

All the best,

David

Digium logo
David Duffett
Digium, Inc. ·  Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green  ·  Bicester, Oxfordshire OX25 3SX  ·  UK
direct/fax:   +1 256 428 6119  · mobile:   +44 7722 442236
twitter:  dduffett   · linkedin:  www.linkedin.com/in/davidduffett  
Check us out at:  http://digium.com  ·  http://asterisk.org  
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[asterisk-users] Asterisk in the London Olympics

2012-09-12 Thread David Duffett

I am looking for any information of the possible use of Asterisk in any of the 
systems used by/for the London Olympics.

Please get in contact if you know of any such use(s).

Thank you,

David


Digium logo
David Duffett
Digium, Inc. ·  Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green  ·  Bicester, Oxfordshire OX25 3SX  ·  UK
direct/fax:   +1 256 428 6119  · mobile:   +44 7722 442236
twitter:  dduffett   · linkedin:  www.linkedin.com/in/davidduffett  
Check us out at:  http://digium.com  ·  http://asterisk.org  
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[asterisk-users] OSS booths at AstriCon

2012-09-06 Thread David Duffett

We have a very small number of booths (3 to be precise) available free*  for 
true OSS projects at AstriCon this year - in our Open Source Solutions Showcase!

*Internet access is not included, and would need to be purchased if required.

Please e-mail me at david.duff...@asterisk.org if you would like to exhibit.

David

Digium logo
David Duffett
Digium, Inc. ·  Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green  ·  Bicester, Oxfordshire OX25 3SX  ·  UK
direct/fax:   +1 256 428 6119  · mobile:   +44 7722 442236
twitter:  dduffett   · linkedin:  www.linkedin.com/in/davidduffett  
Check us out at:  http://digium.com  ·  http://asterisk.org  
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[asterisk-users] As Kevin Fleming says So long, and thanks for all the fish!, we say thank you - and look to the future

2012-07-31 Thread David Duffett



It's amazing what you can learn in a few days...

Having just found out that Queen Elizabeth has a great sense of humor, it has 
now emerged that Kevin Fleming - a man who (both with and without his 
moustache) has been an amazing contributor and influencer in the Asterisk 
project is set to move on to a new challenge outside the project - but still 
within the realms of Open Source.

Kevin has been involved with Asterisk for 7+ years, and has been both a thought 
leader and a powerful voice in the Asterisk world during that time. I first met 
Kevin at a TMC event called VoIP Developer in California (old school, well 
before the days of IT Expo), where he was speaking about Asterisk as well as 
helping to man the Digium booth at the event.

I've also followed Kevin around Berlin looking for great gelato during the 
AstriCon Europe 2006 tour - and it was well worth it, that man knows his gelato!

I'd like to take this opportunity to say thanks to Kevin for his enormous 
contribution to the Asterisk Project. Without his efforts, Asterisk would not 
be the success it is today ... Anyway, back to main theme - when someone in a 
senior role like Kevin moves on, it is important that others are there to pick 
up his responsibilities and move the project forward.

As it turns out, we've already been working on this, and have some very 
talented people that will be taking up the key responsibilities of the project 
going forward. Some of them have been involved with Asterisk for several years, 
and some are recent additions, but together they form a great team to lead 
Asterisk into the future.



Matt Jordan has assumed the project leader role for Asterisk, and is 
responsible for managing the releases of Asterisk, as well as all of the 
development efforts within Digium.
Mark Michelson is serving as the Technical Lead for the project, responsible 
for architecture and design direction.
We have also recently created the role of Community Support Manager, which 
Rusty Newton has filled. Rusty is a long time Digium employee with many years 
supporting Asterisk and Digium products, and will be the day to day interface 
for community technical issues.

As you know, I recently joined Digium to look after the interests of the 
worldwide Open Source Asterisk community and I will therefore also be working 
alongside the good people identified above, especially Rusty.

So while we wish Kevin all the best as he moves on, we are also confident that 
the good work he and the rest of the team have done continues to be in the best 
hands going forwards.

To the future...

David

Digium logo
David Duffett
Digium, Inc. · Director, Worldwide Asterisk Community
6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK
direct/fax: +1 256 428 6119 · mobile: +44 7722 442236
twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett
Check us out at: http://digium.com · http://asterisk.org
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