Hi all,
someone has used the voice recognition software named Sphinx??? Can he tell
me how to use and its performance???
Thanks
Marco
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7 apr 2009 kl. 18.26 skrev Florian Hackenberger:
> On Tuesday 07 April 2009, Olle E. Johansson wrote:
>> I don't see any problems there. YOu still have devices with states,
>> as you would have with authentication. Of course, it still depends on
>> your configuration. But authentication should no
sip show peer ovh
* Name : ovh
Secret :
MD5Secret:
Context : entrant-ovh
Subscr.Cont. :
Language : fr
AMA flags: Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup:
Pickupgroup :
Mailbox :
VM E
I've updated http://www.voip-info.org/wiki/view/Asterisk+func+shell with an
example to test file existence.
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h
I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G phones. Unit obtains SIP trunking
from an ITSP (server has no PCI boards).
*8 Call Pickup works fine from any of the phones connected using the
Linksys SPA2102.
*8 Call Pickup does not
2009/4/7 Mark Michelson
> Philipp Kempgen wrote:
> > Olivier schrieb:
> >> 2009/4/7 Philipp Kempgen
> >>> Olivier schrieb:
> I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext
> in
> >>> an
> AEL2 file like this :
> SendText(${BASE64_DECODE(DQ==)});
>
> >>
hi
i look the list at http://www.voip-info.org/wiki/view/chan_mobile i tryed
whit an N80 and many non listed phones i cant get any one to work to send
sms.
someone know a phone thats really works to send sms? wich one?
thanks!!!
David
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(='.'=)This is Bunny. Copy and paste bunny into you
On Tue, Apr 7, 2009 at 11:54 AM, Gabriel - IP Guys
wrote:
> I have a asterisk setup that is currently running on version 1.4.15 – I wish
> to upgrade or migrate this instance to the current asterisk stable,
> 1.6.0.6. It is my intention to build a FC8 box, then install asterisk from
> source, and
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, April 07, 2009 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall -
canuni
I would upgrade to the latest 1.4, if stable is what is needed.
PaulH
Gabriel - IP Guys wrote:
>
> Dear All,
>
> I have a asterisk setup that is currently running on version 1.4.15 –
> I wish to upgrade or migrate this instance to the current asterisk
> stable, 1.6.0.6. It is my intention to bu
I thought so. Unless someone can write a buffer overrun code to email
them the sip.conf or other config files
then you should be fine if you don't provision unsecured contexts to
dial out to PSTN ...
there was a buffer overrun in chan_sip but it was a couple years ago
Martin
On Tue, Apr 7, 2009
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Tuesday, April 07, 2009 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fwd: [asterisk-r2] MFCR2 With Unicall -
canu
The message includes a host of irrelevant and relevant information. The
question is not clear. It is a horrible piece of top-posting mess.
Please provide the relevant configuration again and clarify your answer.
What hardware do you have? What connections do you have? Are they
working OK?
Gener
Hello there:
I think I have a silly kernel configuration problem. I'm running:
* vanilla 2.6.27.10 kernel built from source
* dahdi-2.1.0.4 built from source
So far so good,
dahdi module loads just fine:
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.1.0.4
when I try to:
ha
And sip set debug peer ovh?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Output of CLI sip show peer ovh?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-use
[ovh]
type=peer
secret=
username=0033972xx
fromuser=0033972xx
host=sip.ovh.net
canreinvite=no
disallow=all
allow=g729
tos_sip=1; Sets TOS for SIP packets.
tos_audio=1 ; Sets TOS for RTP audio packets.
tos_video=1
dtmfmode=rfc28335
relaxdtmf=yes
Max Alex wrote:
> Hi All,
> I have working asterisk version 1.4.24.
> I have a blind transfer issue with grandstream bt200.
Does it work with other phones? That means is it a Grandstream isue or a
general issue?
> I have updated the latest firmware to the phone.
> The phone is sending the *refer
Show us your sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Sent: Tuesday, April 07, 2009 2:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] i have a probleme and my asterisk a
hello every body
my connexion on ovh to pass in UNREACHABLE and not reidentified were not
reboot the server.
[Apr 7 20:17:21] NOTICE[19947]: chan_sip.c:15605
handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms)
[Apr 7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswe
-- Forwarded message --
From: Juan Carlos Huerta
Date: 07-abr-2009 13:41
Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel
commands?
To: asterisk...@lists.digium.com
Please wirte to asterisk-users@lists.digium.com to get help about this
problem.
Juan Carlo
I've just backported it to asterisk 1.4.19,
the patch is atached
2008/8/27 Bob Pierce
>
> On Wed, 2008-08-27 at 11:21 +0300, Atis Lezdins wrote:
> > If you doubt about some part, you're welcome to ask, i'll try to help
> > you, but i don't want to provide complete backport to you, as i won't
>
Philipp Kempgen wrote:
> Olivier schrieb:
>> 2009/4/7 Philipp Kempgen
>>> Olivier schrieb:
I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
>>> an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});
Value sent (8 bytes long) is very strang
On Tue, 7 Apr 2009, Max Alex wrote:
> Hi All,
> I have working asterisk version 1.4.24.
> I have a blind transfer issue with grandstream bt200.
> I have updated the latest firmware to the phone.
> The phone is sending the *refer* to asterisk but asterisk is not able to
> connect with the another c
Olivier schrieb:
> 2009/4/7 Philipp Kempgen
>> Olivier schrieb:
>> > I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
>> an
>> > AEL2 file like this :
>> > SendText(${BASE64_DECODE(DQ==)});
>> >
>> >
>> > Value sent (8 bytes long) is very strange :
>> > Content-Type: text/
It's a bug in the Async AGI feature. I have created a new patch
http://www.moythreads.com/asterisk-1.4.18-async-agi.patch
Please test it and let me know if it works for you,
Moy
On Tue, Apr 7, 2009 at 11:50 AM, Moises Silva wrote:
> "Released" means no patching needed, it means it was tested an
2009/4/7 Philipp Kempgen
> Olivier schrieb:
> > I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
> an
> > AEL2 file like this :
> > SendText(${BASE64_DECODE(DQ==)});
> >
> >
> > Value sent (8 bytes long) is very strange :
> > Content-Type: text/plain;charset=UTF-8
> > Con
Olivier schrieb:
> I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
> AEL2 file like this :
> SendText(${BASE64_DECODE(DQ==)});
>
>
> Value sent (8 bytes long) is very strange :
> Content-Type: text/plain;charset=UTF-8
> Content-Length: 8
>
> �ez?==
I doubt you will
Olivier schrieb:
> Is there any app_backticks
> equivalent or
> workaround for 1.6 ?
SHELL()
http://www.das-asterisk-buch.de/2.1/functions-shell.html
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-
On Monday 06 April 2009 19:22:30 Martin wrote:
> Can you give more information about this vulnerability ?
It's unlikely that it's this vulnerability. Every Asterisk box allows guest
access to the machine, by default. The context it goes to is generally
the "default" context. This is what allows
On Tuesday 07 April 2009, Olle E. Johansson wrote:
> I don't see any problems there. YOu still have devices with states,
> as you would have with authentication. Of course, it still depends on
> your configuration. But authentication should not affect states.
Ok, thanks for that, I'll have a look a
Gabriel - IP Guys wrote:
> Dear All,
>
>
>
> I have a asterisk setup that is currently running on version 1.4.15 – I
> wish to upgrade or migrate this instance to the current asterisk stable,
> 1.6.0.6. It is my intention to build a FC8 box, then install asterisk
> from source, and begin to
Dear All,
I have a asterisk setup that is currently running on version 1.4.15 - I
wish to upgrade or migrate this instance to the current asterisk stable,
1.6.0.6. It is my intention to build a FC8 box, then install asterisk
from source, and begin to migrate over the configuration. The thing is
"Released" means no patching needed, it means it was tested and put
into Asterisk tree. So, I published a patch for 1.4 so it could be
used in 1.4 however the feature per se was just released for Asterisk
1.6.
Moy
On Tue, Apr 7, 2009 at 10:01 AM, wrote:
> Moy,
> I apologize if you felt under so
On Tue, Apr 07, 2009 at 04:28:50PM +0200, Loic Didelot wrote:
> Hello,
> I connected a twinbus system to a xorcom fxs port. I have to set
> immediate=yes to make thins work as expected. And it works.
>
> The problem is that the parameter immediate=yes seems to applied to
> every port, also my PRI
Hi,
I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});
Value sent (8 bytes long) is very strange :
Content-Type: text/plain;charset=UTF-8
Content-Length: 8
�ez?==
Any workaround ?
Regards
__
Hi,
I just figued it out with some help in the irc channel. I just need to
set it back to no for the other channels.
Thanks,
Loic.
On Tue, 2009-04-07 at 16:28 +0200, Loic Didelot wrote:
> Hello,
> I connected a twinbus system to a xorcom fxs port. I have to set
> immediate=yes to make thins work
2009/4/7 mahboob zaman
>
> Hellow
>
> Can any body helps how can interfacing between asterisk and patton media
> getway.
>
Which smartware version ?
>
> Thanks
> mahboob
>
>
> ___
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Hello,
I connected a twinbus system to a xorcom fxs port. I have to set
immediate=yes to make thins work as expected. And it works.
The problem is that the parameter immediate=yes seems to applied to
every port, also my PRI port. This means that extensions are no longer
working for incoming calls.
Hi All,
I have working asterisk version 1.4.24.
I have a blind transfer issue with grandstream bt200.
I have updated the latest firmware to the phone.
The phone is sending the *refer* to asterisk but asterisk is not able to
connect with the another call
that i have checked in sip debug.
I am using
Moy,
I apologize if you felt under some pressure. It wasn't my mind. I only wanted
to know if either there was a mistake in my configuration, or I was failing in
the procedure, or it was a bug, as you said, in order to move forward.
By the way, there's a thing I don't understand:
In your blog a
Greetings listers,
I've posted this at least once previously, but
thought I'd try again. I've got a TDM410P card on Asterisk 1.4.21.2 and
experience these two problems.
1. When placing an outgoing call, I get no audio until Asterisk
bridges the connection (2-15
Hi Enrico,
I do that by modifying logger.conf
[logfiles]
logpro => notice,warning,error,debug,verbose
and modifying asterisk.conf
[directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/li
Here's my .02 - local lan is probably behind a firewall meaning that the
5060 gets out ok to send your audio, but the 1-2 range that the
other side comes in on is blocked. You don't have the problem with static
WAN because it is not behind the firewall or has more ports open. Do a
netstat
Here is a suggestion from the Digium Bug site
asterisk -cvvvgn | tee /tmp/my_log_file.txt
http://bugs.digium.com/view.php?id=14255
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Enrico
Pasqualotto
Sent: Tuesda
2009/4/7 Enrico Pasqualotto :
> Hi all, in witch way can I put in a log file the asterisk console?
> I have tried with some settings in file logger.conf but the log not
> contain the same debug that I can see with "asterisk -rvvv".
> I need it in debugging purpose for tracking some bug.
asterisk -
Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug that I can see with "asterisk -rvvv".
I need it in debugging purpose for tracking some bug.
Thanks Enrico.
smime.p7s
Description: S/MIME c
cyr2...@gmail.com schrieb:
> I'm sorry but I can't find where to change this one in the opensubscriber
> service. I'm sending a request to them for it. As soon as I get the answer
> I''ll do it.
Never mind. I just couldn't resisit. :-)
> -- Philipp Kempgen wrote :
> cyr2...@gmai... schrieb:
>>
7 apr 2009 kl. 11.49 skrev Florian Hackenberger:
> On Tuesday 07 April 2009, Olle E. Johansson wrote:
>> Well, you can have OpenSER doing the authentication and turn it
>> off in Asterisk, but still match a device.
> Ok, but what about sip device state? Will that work? Will asterisk
> report the
7 apr 2009 kl. 12.08 skrev Steve Davies:
> 2009/4/7 Olle E. Johansson :
>>
> [snip]
>>
>> The REGISTER request in the RFC was really written for a device.
>> The way providers use it for trunks with multiple DIDs is outside
>> of the
>> RFC and is discussed in relation to the SIPconnect specifi
Martin escreveu:
Hi,
You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.
The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.
q931.c has this ...
/* For a call in Active sta
Kindly can you send me the code ,or how to
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Monday, April 06, 2009 7:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subj
I'm sorry but I can't find where to change this one in the opensubscriber
service. I'm sending a request to them for it. As soon as I get the answer
I''ll do it.
Regards
Jose Arias
-- Philipp Kempgen wrote :
cyr2...@gmai... schrieb:
> This message was sent on behalf of cyr2...@gmai... at openSu
2009/4/7 Olle E. Johansson :
>
[snip]
>
> The REGISTER request in the RFC was really written for a device.
> The way providers use it for trunks with multiple DIDs is outside of the
> RFC and is discussed in relation to the SIPconnect specification in
> the SIP forum.
>
> Some providers solve this
On Tuesday 07 April 2009, Olle E. Johansson wrote:
> Well, you can have OpenSER doing the authentication and turn it
> off in Asterisk, but still match a device.
Ok, but what about sip device state? Will that work? Will asterisk
report the device as busy when the sip device is engaged in a call?
2009/4/7 Tzafrir Cohen
> On Tue, Apr 07, 2009 at 10:16:41AM +0200, Olivier wrote:
> > Hello,
> >
> > Is there any app_backticks (see
> > http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or
> > workaround for 1.6 ?
>
> That page, while still messy, now references the new
> htt
Hi,
I'm using a SIP phone (Thomson ST2030) which is able to display text
received though Asterisk's SendText() application.
I'm using this to display from Asterisk "Forwarded to 0123456789" whenever a
user forwards his calls to another number or extension.
Test is displayed with white letters on
On Tue, Apr 07, 2009 at 10:16:41AM +0200, Olivier wrote:
> Hello,
>
> Is there any app_backticks (see
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or
> workaround for 1.6 ?
That page, while still messy, now references the new
http://www.voip-info.org/wiki/view/Asterisk+
Ed W wrote:
> Hi, got a Sangoma A200 with a bunch of extension cards and having real
> problems getting it to deal with a normal single BT line
>
> Symptoms are that incoming calls are fine. Outgoing calls ring the far
> end, BUT asterisk never sees that the call is answered (ie no message in
Hello,
Is there any app_backticks (see
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks) equivalent or
workaround for 1.6 ?
In the past, I had trouble trying to use ENV() function.
Cheers
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On Mon, 6 Apr 2009, Ed W wrote:
> Hi, got a Sangoma A200 with a bunch of extension cards and having real
> problems getting it to deal with a normal single BT line
>
> Symptoms are that incoming calls are fine. Outgoing calls ring the far
> end, BUT asterisk never sees that the call is answered (
6 apr 2009 kl. 18.46 skrev Steve Davies:
> Thanks for the reply - Perhaps I was not clear.
>
> On the register=> line, if I set /extension to be /12345, then this
> just replaces 's' with 12345, and ALL calls, regardless of their
> destination number will be routed on the INVITE line to 12...@x.x
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