Re: [asterisk-users] faxdetect on/off on the fly?

...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Thursday, January 03, 2013 3:13 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] faxdetect on/off on the fly? ** ** Hello, We want the ability to choose from an AGI script whether

[asterisk-users] faxdetect on/off on the fly?

Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298

Re: [asterisk-users] faxdetect on/off on the fly?

: On Thu, 3 Jan 2013, David Cunningham wrote: We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. What's the 'use case?' You're going to call in and execute an AGI that will enable faxdetect for future calls to this channel or other

[asterisk-users] Asterisk as a translating proxy only?

format, that may do the trick. Thank you for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] SendFAX timestamp

Steve, Thanks for the reply. Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? On 27 June 2012 00:24, Steve Underwood ste...@coppice.org wrote: On 06/26/2012 10:24 AM, David Cunningham wrote: Hello, Does SendFAX have the ability to put the caller ID

[asterisk-users] SendFAX timestamp

Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642

Re: [asterisk-users] FastAGI script and DIAL execution

: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

Re: [asterisk-users] Continue AGI after Dial() following caller hang up?

to persist for the duration of the call? Cheers, Kingsley. On Tue, 2011-11-22 at 14:27 +1100, David Cunningham wrote: The strange thing is that we are using fast AGI, and for some reason the AGI always exits when the caller hangs up - even when I set HUP to IGNORE. If I set HUP

Re: [asterisk-users] Continue AGI after Dial() following caller hang up?

can catch the signal and do whatever you want to do. Am 21.11.2011 07:38, schrieb David Cunningham: Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h

Re: [asterisk-users] Continue AGI after Dial() following caller hang up?

-info.org/wiki/view/Asterisk+cmd+Dial : F(context^exten^pri): When the caller hangs up, transfer the called party to the specified context and extension and continue execution. Cheers, Kingsley. On Mon, 2011-11-21 at 17:38 +1100, David Cunningham wrote: Hello, We would like to continue a Perl

Re: [asterisk-users] Continue AGI after Dial() following caller hang up?

to the dialplan. I had incorrectly assumed you were doing the same. Cheers, Kingsley. On Mon, 2011-11-21 at 23:01 +1100, David Cunningham wrote: Kingsley, Thanks for the reply, but I am looking to continue within the same AGI process and I believe that method would require starting

[asterisk-users] Continue AGI after Dial() following caller hang up?

. Is this possible? If not a confirmation that this is the case would be very helpful. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

[asterisk-users] Permanent sip and agi debug on?

Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug on in Asterisk? I want this to be automatically enabled even after restarts. Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free

Re: [asterisk-users] Permanent sip and agi debug on?

Kevin, Thank you very much! On 10 November 2011 00:15, Kevin P. Fleming kpflem...@digium.com wrote: On 11/09/2011 04:22 AM, David Cunningham wrote: Hi all, I can't find the answer to this via google - is there some way to permanently enable sip set debug on and agi set debug

[asterisk-users] Voice recognition recommendations?

Hi all, We have a project involving voice recognition, and will need a vocabulary of 10,000 words (actually names). Can anyone recommend a product that works with Asterisk? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642

[asterisk-users] Change to pickups in Asterisk 1.8 - not working on local channels?

-pickup-db70;2' status is 'UNKNOWN' The context doing the pickup looks like: [product-pickup] exten = _[0-9*#]!, 1, Pickup(${EXTEN}@product-phone) Thanks for any advice, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0

Re: [asterisk-users] Higher CPU usage on 1.6.1 than 1.4?

volumes any better. ~Jared On Wed, May 11, 2011 at 8:29 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just

[asterisk-users] Higher CPU usage on 1.6.1 than 1.4?

Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http

[asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module

/drivers/dahdi/dahdi-base.o] Erreur 1 make[1]: *** [_module_/usr/src/dahdi-linux-2.4.0/drivers/dahdi] Erreur 2 make[1]: quittant le répertoire « /usr/src/linux-headers-2.6.34.6 » make: *** [modules] Erreur 2 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0

Re: [asterisk-users] Error compiling Dahdi: invalid use of undefined type struct module

Shaun, CONFIG_MODULES wasn't enabled - thanks for the advice! On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell sruff...@digium.com wrote: On 1/30/11 8:45 PM, David Cunningham wrote: I'm installing Asterisk with Dahdi on a server with a custom kernel compile. I've got the kernel source

[asterisk-users] Introducing easySysAdmin - automated security and telecom fraud protection

. If you have any questions please don't hesitate to contact me directly. Regards, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

[asterisk-users] progressinband, how much extra CPU load?

% increase) that would be great, rather than just lots. Also, are there any ATAs which are known to not work with progressinband = yes? We have Polycom, Linksys and Audiocode. Thanks for any advice, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20

[asterisk-users] 'Bookmarking' a place in a sound file

AGI program. Thanks for any advice! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] 'Bookmarking' a place in a sound file

Steve, that looks just the job, thank you very much. On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 7 Dec 2010, David Cunningham wrote: Is it possible to somehow 'bookmark' a place in a sound file? That is, the user presses a key while a sound file

Re: [asterisk-users] Recording maximum time and stop on silence

Danny, thank you! On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Cunningham *Sent:* Wednesday, September 22, 2010 4:28 PM *To:* Asterisk Users

[asterisk-users] Recording maximum time and stop on silence

All, Two questions: 1. Is there a limit on how long a call can be recorded for? For example is 4 hours a problem? 2. Can recording be stopped after a configured period of silence? Thanks in advance, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0

Re: [asterisk-users] Cause and cure for Exceptionally long voice queue length queuing to Local?

Leif - thank you! Will try that. On Fri, May 21, 2010 at 12:19 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: David Cunningham wrote: Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come

Re: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local?

I don't see anything in the SIP trace related to the warning messages. Would anyone have any further tips? Thanks for any help! On Wed, May 19, 2010 at 9:12 PM, David Cunningham dcunning...@voisonics.com wrote: What should I expect see if it is the peer asking us to slow down RTP? Thanks

[asterisk-users] Cause and cure for Exceptionally long voice queue length queuing to Local?

] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412...@asterisk-phone-7e3d;1 Thanks in advance! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180

Re: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local?

...@lists.digium.com] On Behalf Of David Cunningham Sent: Wednesday, May 19, 2010 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cause and cure for Exceptionally long voice queuelength queuing to Local? Hello, We're seeing lots of warnings like the following

Re: [asterisk-users] What does Asterisk give to reject a re-invite?

:               http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037

[asterisk-users] What does Asterisk give to reject a re-invite?

Hello, If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180

Re: [asterisk-users] What does Asterisk give to reject a re-invite?

have any idea why this is, or where I could go for more information? Thanks for the help. On Thu, May 13, 2010 at 11:06 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/13/2010 01:41 PM, David Cunningham wrote: If you have canreinvite=no and a peer sends you a re-invite, what

[asterisk-users] One way audio problem, a=sendonly and a re-invite

-15. a=sendrecv. a=ptime:20. Any help would be much appreciated! -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180 -- _ -- Bandwidth

Re: [asterisk-users] How to use AGI php script function $agi - exec_dial

by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0

Re: [asterisk-users] Really Silly Question From Total Newbie

list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics Limited IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180

Re: [asterisk-users] Can't load cdr_radius.so module?

CDR record! == Spawn extension (tutorial, 4321, 1) exited non-zero on 'SIP/ivan-0a07dc80' it says Failed to record Radius CDR record. Could you tell me , what's wrong with it? 2009/12/23 Olle E. Johansson o...@edvina.net: 23 dec 2009 kl. 11.25 skrev David Cunningham: Shukun

Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?

, David Cunningham dcunning...@voisonics.com wrote: AsteriskWin32 does have SIP server functionality, same as the linux version. I can't think of any reason why having your CentOS Asterisk be both client and server and register with itself wouldn't work. Although I am wondering how much

Re: [asterisk-users] 1.6 Troubleshooting help

/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180 ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?

list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180

Re: [asterisk-users] Problems with chan_sip

-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia

Re: [asterisk-users] Can't load cdr_radius.so module?

-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0

Re: [asterisk-users] Inquiry:Connect my Asterisk to external sip?

to another SIP provider... On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi motamed...@gmail.comwrote: On Wed, Dec 23, 2009 at 10:22 AM, David Cunningham dcunning...@voisonics.com wrote: Hadi, You could use Asterisk as a sip server, it's installable on Windows. Using sip set debug on might

Re: [asterisk-users] New 1.4 system: registered, but not responding to invite?

://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024

Re: [asterisk-users] SIP to Analog Devices

. Brian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics Limited

Re: [asterisk-users] Best way ro run 2 or more asterisk servers?

To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham Voisonics IVR development, VOIP consultancy http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3411 5024 Australia: +61 (0) 2 9037 2180

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