Hi List,
I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases
hosted on a separate machine). When Asterisk is in verbose mode, it
prints messages saying "failed to extend from 512 to 664" (quite a few
lines in a block) and then the last message is mostly "failed to
extend from 512 to 6
On Fri, 25 Jun 2010 11:19:25 +0100, Gareth Blades
wrote:
>If you are going to reboot the server regularly then make sure and
>system updates are set to not automatically install new kernel versions.
>Otherwise if you get a kernel update and reboot zaptel/dahdi wont load
>until you recompile it.
Hi All,
I am not sure if my comments will be helpful but here goes.
Recently completed a roll out of Asterisk to provide a FTTH (fibre to
the home) based telephone exchange. It uses Asterisk real time and a
MySQL database. The design is for a somewhat smaller audience (~4000
extensions) but shoul
On Fri, 25 Jun 2010 13:25:18 +0300, Tzafrir Cohen
wrote:
>Asterisk 1.4.x works with Zaptel as well.
>
>But yes, this means an upgrade of Asterisk, and maybe you'd like to
>avoid that.
Yup. I'll just stop/start Zaptel every night and see if that fixes the
problem. Thank you.
--
On Fri, 25 Jun 2010 06:45:29 -0700 (PDT), Steve Edwards
wrote:
>However, the "philosophy" of regularly rebooting the box ensures that you
>will never find the source of the problem.
I think I'll just stop/start Zaptel and see if this is the cause for
Asterisk no longer taking call after X weeks.
On Fri, 25 Jun 2010 12:32:03 -0400, Barry Miller
wrote:
>Hi Gilles. You appear to be both posting to newsgroup
>gmane.comp.telephony.pbx.asterisk.user AND sending the same message
>directly to the asterisk-users list. This means that we list subscribers
>see two copies of all your messages: one
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this actio
On Fri, 25 Jun 2010 08:59:32 +0200, Gilles
wrote:
>Before I go ahead, I'd like to know if I can just send the following
>commands, or if there are issues I should know about:
To avoid issues about the host hanging after a reboot due to
upgrades... I think I'll just run a CRON job to stop/start Za
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Elvish
Sent: Saturday, June 26, 2010 6:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Big time system
Hi
Hello
Googling for this type of non-PC hardware returns products that could
be missing in action for years.
www.google.com/search?q=asterisk+appliance
Is there an up-to-date list of Asterisk appliances, ideally broken
down by price (ie. not just entreprise stuff, but also SOHO)?
Thank you.
--
Hello
To run Asterisk on an embedded appliance, ie. where RAM and
non-volatile memory is an issue (respectively 64MB and 256MB), I need
to check how much space voice messages take to save and play back.
The appliance is connected to a landline in Europe (in case that makes
a difference as
Hello,
When I call "dialplan reload" I can see the following lines:
== Parsing '/etc/asterisk/extensions.conf': == Found
-- Registered extension context 'default' (0x8a72410) in local table
0x8a679d0; registrar: pbx_config
-- Added extension '_1XX' priority 1 to default (0x8a72410)
.
On Sat, 26 Jun 2010, Gilles wrote:
> Hello
>
> To run Asterisk on an embedded appliance, ie. where RAM and
> non-volatile memory is an issue (respectively 64MB and 256MB), I need
> to check how much space voice messages take to save and play back.
>
> The appliance is connected to a landline
Honest-to-god CNAM dip:
/CNAM/9162198187 : TELEMARKET SPAM
(This is the actual CallerID Name as dipped from AT&T CNAM this morning)
ROTFL
-Karl
--
_
-- Bandwidth and Colocation Provided by htt
On Sat, Jun 26, 2010 at 7:33 AM, Deepesh D wrote:
> Is it possible to do this action on hook flash?
>
Currently no. You would need to add logic to the channel driver. Or
use DTMF to initiate the hookflash:
extensions.conf
[globals]
DYNAMIC_FEATURES=>zapflash
features.conf
[applicationmap]
zapf
On Sat, Jun 26, 2010 at 8:25 AM, Gilles wrote:
> Is there an up-to-date list of Asterisk appliances, ideally broken
> down by price (ie. not just entreprise stuff, but also SOHO)?
>
Might get better results on asterisk-biz, and posting your budget price range.
--
Paul Belanger | dCAP
Polybeacon
On Sat, Jun 26, 2010 at 12:43 PM, Eyal Goltzman wrote:
> I don't want those 2 first line to be there only the _1XX. How do I get rid
> of it? Why it is added to the dial plan automatically?
>
Stop using users.conf. You will have more control using sip.conf and
extension.conf.
--
Paul Belanger |
On Sat, Jun 26, 2010 at 9:21 AM, Gilles wrote:
> To run Asterisk on an embedded appliance, ie. where RAM and
> non-volatile memory is an issue (respectively 64MB and 256MB), I need
> to check how much space voice messages take to save and play back.
>
Doing some simple math and using the se
On Saturday 26 June 2010 04:28:18 Stuart Elvish wrote:
> I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases
> hosted on a separate machine). When Asterisk is in verbose mode, it
> prints messages saying "failed to extend from 512 to 664" (quite a few
> lines in a block) and then the l
Hello,
After installing and learning Asterisk I found myself with a need for a
minimal set of empty configuration files with only the "must have" stuff in
order to setup a SIP only machine, is there a place to find it?
Thanks,
Eyal
--
__
On Sat, Jun 26, 2010 at 2:09 PM, Eyal Goltzman wrote:
> Hello,
>
> After installing and learning Asterisk I found myself with a need for a
> minimal set of empty configuration files with only the "must have" stuff in
> order to setup a SIP only machine, is there a place to find it?
Depends on how
Hello,
I use cri
http://www.tikalnetworks.com/voip/index.php?cid=38
Best regards
On Thu, Jun 24, 2010 at 3:22 AM, Mickael Monsieur <
mickael.monsi...@gmail.com> wrote:
> Hello Bruce,
>
> This module is not reliable on FreePBX?
> You know if there is a open source web-voicemail for Asterisk?
>
On Wed, Jun 23, 2010 at 12:57 PM, James Lamanna wrote:
> On Tue, Jun 22, 2010 at 8:57 PM, Andres wrote:
>>
>>> completely as well.
>>>
>>> Below I've posted a patch that responds with a 200 OK to these
>>> keep-alive requests, and I believe
>>> also solves the temporary loss of registration probl
I have Polycom phones that support the g722 codec. Adding allow=g722
to the [general] section of sip.conf works great and I can make calls
between the phones using g722. However Asterisk is negotiating g722
for calls going out my voip provider and transcoding these to ulaw. In
sip.conf for the prov
On Sat, Jun 26, 2010 at 05:53:27PM +0100, Gordon Henderson wrote:
> On Sat, 26 Jun 2010, Gilles wrote:
>
> > Hello
> >
> > To run Asterisk on an embedded appliance, ie. where RAM and
> > non-volatile memory is an issue (respectively 64MB and 256MB), I need
> > to check how much space voice mes
On Sat, 26 Jun 2010 21:09:54 +0300, "Eyal Goltzman"
wrote:
>After installing and learning Asterisk I found myself with a need for a
>minimal set of empty configuration files with only the "must have" stuff in
>order to setup a SIP only machine, is there a place to find it?
The experts might chime
Does it exist? Sending email to their support address appears to be a
black hole. They reference a forum, but Google can't find it.
I keep having problems in any grammar than has a an "o" for "zero": it breaks
recognition anywhere NEAR it. For example, if I say "two o five", it gets
recognized
On Sat, 26 Jun 2010 13:35:12 -0400, Paul Belanger
wrote:
>Might get better results on asterisk-biz, and posting your budget price range.
I'll check it out. Thanks Paul
--
_
-- Bandwidth and Colocation Provided by http://www.ap
I want a call to connect via my DID to my dialplan.
Then, I want to attach a label to the incoming call
call arrives
starts to dive through the dial plan
then rings a trunk/channel via SIP (see below)
Question: before answering my 1212111 endpoint I want to see a
flag CID that correlates to th
exten => s,n,Set(CALLERID(name)=label${CALLERID(name)})
put this before the dial command.
On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron wrote:
> I want a call to connect via my DID to my dialplan.
> Then, I want to attach a label to the incoming call
>
> call arrives
> starts to dive through th
ok
thank you
i will try
On Sat, Jun 26, 2010 at 10:31 PM, C F wrote:
> exten => s,n,Set(CALLERID(name)=label${CALLERID(name)})
> put this before the dial command.
>
> On Sat, Jun 26, 2010 at 10:09 PM, Thomas Perron
> wrote:
>> I want a call to connect via my DID to my dialplan.
>> Then, I wan
Hi Richard,
Reference your query, Vestec provides free
installation support for its speech recognition engine.
As
stated on our website at http://www.vestec.com/support_overview in order to
post a tech support request, you first need to send an
email to supp...@vestec.com You will then receiv
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