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
omer:2] Set("SIP/139255423-004c",
> "CDR(userfield)={"agi":"","a-leg-id":"2118d872-305e-4bb4-8c47-30e1514cb934","b-leg-id":"36b232e73ac326bd0407b1594627c589@y.y.y.y:5060"}")
> in new stack
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 wrote:
> On 06/26/2012 10:24 AM, David Cunningham wrote:
>
>> Hello,
>>
>> Does SendFAX have the ability to put t
Kevin,
Thanks for the reply.
On 29 June 2012 00:29, Kevin P. Fleming wrote:
> On 06/27/2012 09:30 PM, David Cunningham wrote:
>
> Would anyone else know if Asterisk allows use of SpanDSP's time zone
>> conversion?
>>
>
> No, SendFAX (in res_fax) doesn't c
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
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
ew to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mai
um.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 AG
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
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
Kevin,
Thank you very much!
On 10 November 2011 00:15, Kevin P. Fleming 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 o
ot for caller hangup.
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:
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
> us
w.voip-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:
> > going into the h extension.
> >
> > I'm not doing any AGI stuff here that hangs around while the call does
> stuff
> > - the AGI process just runs quickly then quits, returning control back to
> > the dialplan. I had incorrectly assumed you were doing the same.
&
you want your fastAGI instance 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 wh
g "30% 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: +
ed easySysAdmin.
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
Austr
2]: *** [/usr/src/dahdi-linux-2.4.0/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
Shaun,
CONFIG_MODULES wasn't enabled - thanks for the advice!
On Mon, Jan 31, 2011 at 4:02 PM, Shaun Ruffell 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&#
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
o see if it handles large call 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
>> expe
-- Auto fallthrough, channel 'Local/1000103@product-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
he event I am totally wrong about whether
> this would actually work, suggest alternatives?
>
> Many thanks.
>
> Brian
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users
m one end to the other I get:
>
> [Dec 15 03:06:04] NOTICE[4265]: chan_iax2.c:10338 socket_process: Host
> 172.16.0.3 failed to authenticate as 300
>
>
> Please help.
>
> Thanks.
>
>
>
>
> ___
> -- Bandwidth and
stem) the called phone does
> >> not ring.
> >>
> >> Has anyone bumped into this lately?
>
>
> ___
> -- 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
>>
>
>
>
> Thank you for your reply . Please be informed that I want to simulate this
> case in the Laboratory , i.e. connecting my Asterisk sip to external sip
> server w
;
> What is going on here ??
>
> Jonas.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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t; Regards,
> Sucan
>
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>
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>
problem to another SIP provider...
On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi wrote:
>
>
> 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 instal
ERROR[10747]: cdr_radius.c:227 radius_log: Failed to
> record Radius 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?
&g
ith itself wouldn't work.
>> Although I am wondering how much help all this will be in debugging a
>> connection problem to another SIP provider...
>>
>>
>> On Wed, Dec 23, 2009 at 12:55 PM, hadi motamedi wrote:
>>
>>>
>>>
>>> On W
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>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asterisk-u
gt; ___
> >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> >
> >> > asterisk-users mailing list
> >> > To UNSUBSCRIBE or update options visit:
> >> > http:/
can
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
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> http://lists.digium.com/mailman/listinfo/asteri
ap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
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
--
__
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
called party can't hear the caller.
Do you 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 wrote:
> On 05/13/2010 01:41 PM, David Cunningham wrote:
>
>> If you have canreinvite=no and
Hi Kevin,
We don't have "mohinterpret" set at all, so I think it uses "default".
Is there anything else you can suggest? Any other places to go for
help?
Thanks for your assistance!
On Thu, May 13, 2010 at 11:32 PM, Kevin P. Fleming wrote:
> On 05/13/2010 05:1
7 13:42:45] 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
.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,
>
&
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
wrote:
> What should I expect see if it is the peer asking us to slow down RTP?
>
> Thanks again.
>
Leif - thank you! Will try that.
On Fri, May 21, 2010 at 12:19 AM, Leif Madsen
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?
>>
>>
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
Danny, thank you!
On Wed, Sep 22, 2010 at 10:31 PM, Danny Nicholas 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
hin a perl 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 Provid
Steve, that looks just the job, thank you very much.
On Wed, Dec 8, 2010 at 2:32 AM, Steve Edwards wrote:
> 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 so
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