Hello,
I had posted this mail some time back, Having got no responses I tried
one suggestion I received in another thread and replaced all IAX
trunks with SIP trunks. That has resolved this issue. Asterisk now
does not hit more than 100% CPU and there is no call disturbance. CPU
usage is now is mo
I have done something similar using the following:
1. An Adit 600 with FXS card.
2. A door box from Viking
http://vikingelectronics.com/products/view_product.php?pid=428
3. An inline dialer from viking:
http://vikingelectronics.com/products/view_product.php?pid=137
4. A relay activated using an FXS
Sorry but AIPHONE is a terrible choice for this.
On Thu, Sep 24, 2009 at 8:53 AM, Chris Mason (Lists) wrote:
> AIPHONE makes all that stuff, I would not try to reinvent that.
>
> Vincent wrote:
>> Hello
>>
>> I assume I'm not the first one to think about this: Is it possible to
>> connect an inte
On Thu, 2009-09-24 at 09:56 -0400, jon pounder wrote:
> Dean Collins wrote:
>
> Earlier in the thread someone made a comment about using gsm since
> everyone had gsm handsets already.
>
> Can you explain in detail please ? (what hardware specifically, and how
> does this actually work ?) My ign
I've tried turning logging way up for the relevant portions of the sip
application, but no telnet. Not sure how I would go about this to get more
info that what I already have. The phone is giving me a response, it's just
that the response
is "push message cannot be displayed"
Mike
> -Origi
Hi,
yes I did, I did have errors at first but that hurdle has been cleared.
Thanks for the try :-)
Mike
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Dave Fullerton
> Sent: Thursday, September 24, 20
On Thu, 2009-09-24 at 16:20 +0300, Tzafrir Cohen wrote:
> On Thu, Sep 24, 2009 at 02:47:18PM +0200, Vincent wrote:
> > Hello
> >
> > I assume I'm not the first one to think about this: Is it possible to
> > connect an intercom and/or door bell to Asterisk, so that I can get an
> > e-mail that some
On Thu, Sep 24, 2009 at 05:32:24PM -0500, Michael Graves wrote:
> On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote:
>
> >Hi,
> >
> >Given that the Digium transcoding card has no external connections
> >(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
> >
> >Does such a be
On Thu, 24 Sep 2009 09:42:25 +0100, Steve Davies wrote:
>Hi,
>
>Given that the Digium transcoding card has no external connections
>(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
>
>Does such a beast exist, or is it likely to? Am I correct in assuming
>that this is a Digium-o
sean darcy wrote:
> Martin wrote:
>> if you're trying to send the same fax to both parties, then do
>>
>> exten => s,1,System()
>> exten => s,2,Sendfax()
>>
>> step1 will spool the call to dial a number and send a fax
>> step2 will transmit the fax to the incoming call
>>
>> Martin
>>
>> On Wed, Se
Double Grrr... I have a NING ID, but no invite.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, September 24, 2009 2:50 PM
To: randulo2...@gmail.com; Asterisk Users Mailing List -
On Thu, 24 Sep 2009, randulo wrote:
> Take a look at this:
>
> http://food4wine.ning.com/forum/topics/submit-an-application-for
Grrr.
Have to have a "Ning" ID and you have to be "invited."
--
Thanks in advance,
-
Steve Edw
Hi,
Take a look at this:
http://food4wine.ning.com/forum/topics/submit-an-application-for
Way down the page Dave VG submitted some scripts that hold the answers.
We also did a Polycom App conference at the VUC, but I can't find the
link right now.
/r
__
Chandrakant Solanki wrote:
> Hi
>
> r u forwarding call using Originate action..
>
> Which version of asterisk u used.
>
Hi
asterisk 1.6.2.0
I'm using freepbx, but I looked into the generated files: if I read it
correctly it ends up using Dial cmd.
thanks,
John
__
In case it's important to you, microbrowser support was added to the 501
and 430 back in SIP 2.1.0. Though how you could use a microbrowser on a
430 for much I don't know.
-Dave
Danny Nicholas wrote:
> This is also a stab-in-the-dark as my 501 doesn't have a microbrowser; Have
> you tried com
I am using the 3Com Unified Gigabit Wired and Wireless PoE Switch. See
the link below.
http://www.3com.com/products/en_US/detail.jsp?tab=features&pathtype=purchase&sku=3CRUS2475
Jason Baker
IT
Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4
This is also a stab-in-the-dark as my 501 doesn't have a microbrowser; Have
you tried communicating with the phone via telnet to debug the problem?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Mike wrote:
> Hi,
>
>
>
> I have been trying a (really simple) push application for the Polycom
> microbrowser, using a Polycom 650 with 3.2 firmware.
>
>
>
> I can't do anything, I always get "Push message cannot be displayed" back
> from the Polycom phone, and all I am sending is the Poly
Greetings,
We'll be getting together as usual at 12 Noon Eastern US Time for a
chat with David Duffet, a well-known member of the Asterisk community
and hopefully one or more of his co-authors of the new book Asterisk
1.4 Professionals Guide. In fact, I've been offered two ebook version
to give aw
Hi,
I have been trying a (really simple) push application for the Polycom
microbrowser, using a Polycom 650 with 3.2 firmware.
I can't do anything, I always get "Push message cannot be displayed" back
from the Polycom phone, and all I am sending is the Polycom example :
Fire Drill a
Yes - I have a similar access control using VoIP Pantel (Aleen) and Viking
Units w- a C1000 module
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent
Sent: Thursday, September 24, 2009 8:47 AM
To: asterisk
On Thu, 24 Sep 2009, Vincent wrote:
> Hello
>
> I assume I'm not the first one to think about this: Is it possible to
> connect an intercom and/or door bell to Asterisk, so that I can get an
> e-mail that someone rang my place while I was out?
>
> Even better: If used for a doctor's office, it'd b
What unit is dtmftimeout measured in? The sample configuration is
provided below. Does it mean to say that the sample configuration
file's dtmftimeout=3000 equates 1/8000th of a second?
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/800
Martin wrote:
> if you're trying to send the same fax to both parties, then do
>
> exten => s,1,System()
> exten => s,2,Sendfax()
>
> step1 will spool the call to dial a number and send a fax
> step2 will transmit the fax to the incoming call
>
> Martin
>
> On Wed, Sep 23, 2009 at 7:45 PM, sean
haven't heard of Digium miniPCI transcoding card ... but who knows
maybe they're working on it ...
Martin
On Thu, Sep 24, 2009 at 3:42 AM, Steve Davies wrote:
> Hi,
>
> Given that the Digium transcoding card has no external connections
> (AFAIK), it strikes me that it would suit a mini-PCI slot
On Thu, Sep 24, 2009 at 8:47 AM, Vincent wrote:
> I assume I'm not the first one to think about this: Is it possible to
> connect an intercom and/or door bell to Asterisk, so that I can get an
> e-mail that someone rang my place while I was out?
>
Valcom makes a SIP door phone, but they're fairly
On Thursday 24 September 2009 05:06:02 stephen.hindma...@bt.com wrote:
> I have investigated further and found that it is a bug in ODBC, not
> Asterisk. The SQLColumns function, which asterisk uses to describe the
> table, does not return any columns when the table name includes the
> schema specif
Dean Collins wrote:
Earlier in the thread someone made a comment about using gsm since
everyone had gsm handsets already.
Can you explain in detail please ? (what hardware specifically, and how
does this actually work ?) My ignorant assumption is something like the
end user has a cell phone th
Sriram escribió:
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100
& 101 ) in a queue..When a caller arrives in queue , it lands on
first 100 , 100 then does a blind transfer to 101 .. so that the
caller can converse with 101 .. strangely enough the queue_log
if you're trying to send the same fax to both parties, then do
exten => s,1,System()
exten => s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy wrote:
> Martin wrote:
>> we
just forget about the dial(a,G()) approach ... you already posted that
it doesn't work ...
either call sendfax on the 1st step
to send fax to the channel that called in to asterisk or
use that call to trigger sending a fax with originate/system
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy
DECT rocks - I understand the reasons for wanting to use wifi but
sometimes when it's raining it makes more sense to drive a motorcar
instead of ride a motorcycle :-)
Cheers,
Dean
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-u
On Thu, Sep 24, 2009 at 02:47:18PM +0200, Vincent wrote:
> Hello
>
> I assume I'm not the first one to think about this: Is it possible to
> connect an intercom and/or door bell to Asterisk, so that I can get an
> e-mail that someone rang my place while I was out?
>
> Even better: If used for a d
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
1253814090
Hi ,
I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101
) in a queue..When a caller arrives in queue , it lands on first 100 , 100
then does a blind transfer to 101 .. so that the caller can converse with
101 .. strangely enough the queue_log shows :
1253814090|12538
Just out of curiosity, what managed switch you used on this project?
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- "Jason Baker" escreveu:
> I think that if I could go back and do this project over, I would have
> chosen DECT as well. W
AIPHONE makes all that stuff, I would not try to reinvent that.
Vincent wrote:
> Hello
>
> I assume I'm not the first one to think about this: Is it possible to
> connect an intercom and/or door bell to Asterisk, so that I can get an
> e-mail that someone rang my place while I was out?
>
> Even be
Hello
I assume I'm not the first one to think about this: Is it possible to
connect an intercom and/or door bell to Asterisk, so that I can get an
e-mail that someone rang my place while I was out?
Even better: If used for a doctor's office, it'd be cool if patients
could type their Social Securi
I think that if I could go back and do this project over, I would have
chosen DECT as well. We have intermittent problems with the wifi AP's
also.
Jason Baker
IT
Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
Fax: 989.752.4276
w
On Thu, Sep 24, 2009 at 10:54:17AM +0200, Loic Didelot wrote:
> Not sure,
> how can I check, but older astribanks work pretty fine on that system.
ls /proc/bus/usb
What is the output of:
lsusb
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7
I have investigated further and found that it is a bug in ODBC, not
Asterisk. The SQLColumns function, which asterisk uses to describe the
table, does not return any columns when the table name includes the
schema specification. You can show this by using isql to do "help table"
which returns info
Not sure,
how can I check, but older astribanks work pretty fine on that system.
Loic
On Wed, 2009-09-23 at 15:00 +0300, Tzafrir Cohen wrote:
> On Wed, Sep 23, 2009 at 08:37:19AM +0200, Loic Didelot wrote:
> > Hi Tzafrir,
> > I just compiled the tarball, but now there seem to be some problems wi
Hi,
Given that the Digium transcoding card has no external connections
(AFAIK), it strikes me that it would suit a mini-PCI slot very well.
Does such a beast exist, or is it likely to? Am I correct in assuming
that this is a Digium-only product, and there is no OEM equivalent
"generic" board out
Hi
r u forwarding call using Originate action..
Which version of asterisk u used.
On Thu, Sep 24, 2009 at 12:44 PM, John Fawcett wrote:
> In some circumstances I am transferring incoming calls to an external
> number (cell phone). Whenever this happens at the end of the call I get
> a single C
In some circumstances I am transferring incoming calls to an external
number (cell phone). Whenever this happens at the end of the call I get
a single CDR representing the outgoing leg. There is no CDR for the
incoming leg and no trace of incoming caller id in the CDR for outgoing
leg.
Is this exp
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