On Wed, 2011-01-19 at 11:41 -0600, Jason Parker wrote:
On 01/19/2011 04:41 AM, Ishfaq Malik wrote:
Hi
Does anyone have any idea how long it will take for the new release of
asterisk 1.8 to make it to the digium yum repositories?
Thanks
Ish
They've been there since yesterday
How amusing that you follow that statement by being too lazy to trim
all of the irrelevant crud after your comment by pressing
ctrl-shift-end followed by delete. It works in Outlook.
Tom
This is the problem, everyone has a personal goal. One side wants fast
replies at the top, with no
On Jan 19, 2011, at 11:08 PM, DSR wrote:
Is there anyway to play prerecorded agent intro-speech (like Hello, my name
is ) to outside caller when agent picks up?
I don't know of a way to do that, but I can say that, as a caller, it is highly
annoying. Your agents ought to be able to do
On 20 January 2011 11:54, Tom Rymes try...@rymes.com wrote:
I don't know of a way to do that, but I can say that, as a caller, it is
highly annoying. Your agents ought to be able to do that themselves, no?
Exactly, otherwise you are losing first chance to make the call different
from the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Thursday, January 20, 2011 3:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Top Posting
How
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bart
Swedrowski
Sent: Thursday, January 20, 2011 6:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hi, agent intro-speech for
Hi all,
I realize that the application Receivefax can't handle with more than one fax
at the same time. In a environment with a lot of fax, some caller get the
signal but the operation can't be completed. Is there a way to send busy tone
to the second caller?
Att,
Flavio Roberto
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs for faxing.
Is see Application_SendFax and Application_SendeFax has one been
discondinued?
On 01/20/2011 09:00 AM, Flavio Miranda wrote:
Hi all,
I realize that the application Receivefax can't handle with more than
one fax at the same time. In a environment with a lot of fax, some
caller get the signal but the operation can't be completed.
Is there a way to send busy tone to the
Jonas Kellens wrote:
[snip]
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[TRUNKin]
exten = _52525252,1,NoOp(context TRUNKin - 52525252)
exten = _52525252,n,GoTo(blabla,52525252,1)
exten = _59595959,1,NoOp(context TRUNKin - 59595959)
exten =
Hi,
For an organization welcoming turists (in France), I would be curious to
learn about successful use (with Asterisk) of Text-To-Speech in spanish (and
english).
I took a look at Cepstral's web site and saw there 2 Americas Spanish
voices (along a bunch of english voices).
1. In this context,
Hi,
I set up ReceiveFax to answer a specific number (2134-4805) , so , the first
caller get the fax signal and transmit the fax normal, but, if another caller
to call the same number almost at the same time, it gets the signal as well but
the fax is not sent!
Att,
Flavio Roberto
On 01/20/2011 04:43 PM, Jose P. Espinal wrote:
Jonas Kellens wrote:
[snip]
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[TRUNKin]
exten = _52525252,1,NoOp(context TRUNKin - 52525252)
exten = _52525252,n,GoTo(blabla,52525252,1)
exten =
On Thu, 20 Jan 2011, Danny Nicholas wrote:
All Asterisk prompts are configurable with a little legwork. Simply use
the CLI to see what is playing at the point you want to change, then set
up this little ditty to override it. Say you wanted to record the
“canned” tt-weasels prompt (“Weasels
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, January 20, 2011 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT - TTS in spanish
Hi,
For an
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, January 20, 2011 9:00 AM
To: Asterisk Asterisk
Subject: [asterisk-users] ReceiveFax
Hi all,
I realize that the application Receivefax can't
On 01/20/2011 04:29 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, January 20, 2011 9:20 AM
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten =
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, January 20, 2011 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] context problem
On 01/20/2011
This is new to me, I have a fax server using Receive Fax and gets way over 5
calls at a time.
[fax-in]
exten = s,1,Answer()
exten = s,n,Wait(1)
exten =
s,n,Set(BASEFILE=fax-${CALLERID(dnid)}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
;exten = s,n,Set(${LOCALSTATIONID})
exten =
Hi,
I know you can access various sip variables via
'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of
the sip user - but what about variables?
I have a user that has setvar=123456 in their users.conf (sip.conf if
you prefer). I can read it with a 'sip show peer
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Thursday, January 20, 2011 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Accessing a
On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote:
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten =
On 01/20/2011 11:00 PM, Flavio Miranda wrote:
Hi all,
I realize that the application Receivefax can't handle with more than
one fax at the same time. In a environment with a lot of fax, some
caller get the signal but the operation can't be completed.
Is there a way to send busy tone to
From: William Stillwell will...@stillwellsoft.com
Sent: Thursday, January 20, 2011 11:26 AM
This is new to me, I have a fax server using Receive Fax and gets way over 5
calls at a time. [fax-in] exten = s,1,Answer() exten = s,n,Wait(1) exten
=
That's what I am already using :)
Somehow, the outbound ID sometimes gets messed up (maybe to do with 2
calls from different users at once) - and the wrong one is sent to the
telco.
So, rather than just using a 'Set(CALLERID(num)=callidnum' just before
Dial - I wanted to check the user directly
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs for faxing.
Is see Application_SendFax and
I always thought the last bit (after the /) is where the context in
sip.conf landed.
What about:
(sip.conf)
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[52525252]
...
context = TRUNKin52
...
[59595959]
...
context = TRUNKin59
...
And
Hi,
Is the any kind of 'tag' that I can include at the end of my message to
make the list processing software ignore and dispose of my disclaimer?
In other words - something like disclaimer at the end of my message
would inform the list software to remove any lines after it.
My massive
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Thursday, January 20, 2011 11:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Mailing list
On 01/20/2011 12:01 PM, Andrew Thomas wrote:
why not just subscribe with an account that doesn't do that like gmail
or yahoo ?
Hi,
Is the any kind of 'tag' that I can include at the end of my message to
make the list processing software ignore and dispose of my disclaimer?
In other words -
Let's see :)
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 20 January 2011 17:04
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Mailing
That's my last option Jon.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon
pounder
Sent: 20 January 2011 16:59
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Mailing list question
On
On 01/20/2011 10:58 AM, Jonas Kellens wrote:
[snip]
I have the following registrations :
register = 119909:pas...@sip.prov.org/52525252
mailto:119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
mailto:119909:pas...@sip.prov.org/59595959
[snip]
Problem :
Sorry about this - testing this disclaimer problem :)
--
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the
Is the any kind of 'tag' that I can include at the end of my message
to make the list processing software ignore and dispose of my
disclaimer?
It looks like there were underscores on the same line as the --
I think the actual idea is to include '-- ' with nothing else on that line
--Don
--
On 20 Jan 2011, at 17:13, Andrew Thomas wrote:
Sorry about this - testing this disclaimer problem :)
I can give you a POP3 account on my server if it stops you spamming the list?..
S
--
_
-- Bandwidth and Colocation Provided by
On Thu, Jan 20, 2011 at 12:03 PM, Danny Nicholas da...@debsinc.com wrote:
Putting the -- in front of it might make it go away.
If I am not mistaken it should be exactly
two dashes followed by a space on a line alone
to indicate the end of the mail content.
But not all mail readers will honor it.
Sorry Dannny - it didn't work :(
I can only hope that someone at API has the answer.
Thanks for trying :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 20 January 2011 17:04
To:
Tell you what Steve - I'll not take you up on your kind offer - I'll
just let my server keep adding the disclaimer.
There - problem solved.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 20
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing re-invites?
--
Hi,
Tomorrow, our discussion is around iNum with lots of interesting
people chiming in, including the Voxbone people who manage the space.
If you ever wondered about iNum and why you might care about it, how
it works, who offers it and who actually uses it, here's a chance to
find out more.
Join
All,
I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware
2.1.3. I can not seem to get the Message Waiting Indicator to work
reliably (and in my opinion correctly) with voicemail.
I've got the following in my phone.cfg:
reginfo
msg msg.bypassInstantMessage=1
mwi
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then 'asterisk -vr' to attach to the
asterisk console there are no colors. If I use the safe_asterisk
script to start
On 20 January 2011 18:01, JR Richardson jmr.richard...@gmail.com wrote:
Or is there another work around to get ssh console colors using the
Debian * 1.6.0.28 LSB init script?
I also tried 'nocolor = no' in the [options] section of asterisk.conf
with no effect.
Try running asterisk using
Brian C. Huffman wrote:
Does anyone know how to setup this phone to work with asterisk so that
the indicator light comes on when there's a new message and goes off
quickly (less than a minute) after the message is deleted?
My phone.cfg for extension 4221 and the voicemail extension of 4200
On Thursday 20 Jan 2011, JR Richardson wrote:
Hi All,
I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the
asterisk daemon not the safe_asterisk daemon so when asterisk is
running and I ssh tot he server then 'asterisk -vr' to attach to the
asterisk console there are no
On 01/20/2011 11:16 AM, Andrew Thomas wrote:
Sorry Dannny - it didn't work :(
I can only hope that someone at API has the answer.
Thanks for trying :)
API provides the physical services and bandwidth for the mailing lists,
but does not operate them. If you go to the lists.digium.com site
On Thu, Jan 20, 2011 at 12:55 PM, Brian C. Huffman
bhuff...@etinternational.com wrote:
Does anyone know how to setup this phone to work with asterisk so that the
indicator light comes on when there's a new message and goes off quickly
(less than a minute) after the message is deleted?
Thanks,
I may be wrong here, but I think you can only register once. The last
registration received will overwrite the first one. You will need to
specify a second entry and register that one separately. This is the
same reason you cannot register two devices to the same extension.
Yes, that's
I understood that option worked the other way around so attacker
thinks peer name is invalid even when they hit a real one.
On Wed, Jan 19, 2011 at 2:23 AM, ad...@3a.hu wrote:
Hi List,
i've been receiving several sip registration probes in the last month, and
as this server is a testing site
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I
can send recieve faxes from both boxes fine to and from pstn. But the
faxing between 1.6 and 1.4 extensions does fail. Any ideas please ?
--
Thank You
Amit Nepal
--
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal ami...@phoenixinternet.net wrote:
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can
send recieve faxes from both boxes fine to and from pstn. But the faxing
between 1.6 and 1.4 extensions does fail. Any ideas please ?
You
I've got the following in my phone.cfg:
reginfo
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.callBack=*97 msg.mwi.1.callBackMode=contact
msg.mwi.1.subscribe= /mwi
/reginfo
The actual config looks good, but the structure of the XML is off. Here's
what I use (and it works):
phone1
msg
The Asterisk Development Team has announced a release for the security issue
described in AST-2011-001.
Due to a failed merge, Asterisk 1.8.2.1 which should have included the security
fix did not. Asterisk 1.8.2.2 contains the the changes which should have been
included in Asterisk 1.8.2.1.
Hi Kyle,
On 01-20-2011 20:41, Kyle Kienapfel wrote:
I understood that option worked the other way around so attacker
thinks peer name is invalid even when they hit a real one.
sorry, it must be because i'm not a native english speaker but i don't
exactly get what you mean by the above.
to
Hi,
I have an Audio code gateway between two asterisk servers. The
audio code has PRI connected for PSTN. I can send faxes and receive
faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN)
and receive faxes. The only problem I am having is sending/receiving
between ast
Hello all,
Voisonics is pleased to introduce easySysAdmin, an automated
support/security platform, designed to save your engineer's time and prevent
hacking attempts and telecom fraud.
It comprises of an online service run by us, and a lightweight and
easy-to-install client on your side.
Sorry, this got buried in my inbox. Did you find a fix or the firmware?
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jfratant...@iswan.net
Sent: Friday, January 14, 2011 6:05 PM
To: Asterisk Users
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can send faxes and receive faxes in
ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and
receive faxes. The only problem I am having is
On 01/20/2011 11:47 AM, Steve Underwood
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs for
Yes Tom,
I am sending via the PSTN gateway which is audio code in my case.
Thank You
Amit Nepal
On 1/20/2011 3:07 PM, Tom Rymes wrote:
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can
On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote:
On 1/20/2011 3:07 PM, Tom Rymes wrote:
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can send faxes and receive faxes in
ast 1.4 . Also I can
Amit
Make sure that the trunk you have between the two servers has the t.38
enabled on it. Do you have any NAT between the two servers or are they on
the same lan. We do the t.38 faxing between 1.4 and 1.6 asterisk boxes all
of the time. Our audio codes gateway dumps into a 1.4 box and all
On 01/20/2011 03:29 PM, David Cunningham wrote:
Hello all,
Voisonics is pleased to introduce easySysAdmin, an automated
support/security platform, designed to save your engineer's time and
prevent hacking attempts and telecom fraud.
As the description of this mailing list says, it is for
On 01/21/2011 06:46 AM, Bryant Zimmerman wrote:
On 01/20/2011 11:47 AM, Steve Underwood
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am
On Jan 20, 2011, at 8:53 PM, Steve Underwood
On 01/21/2011 06:46 AM, Bryant Zimmerman wrote:
On 01/20/2011 11:47 AM, Steve Underwood
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am
On Thu, Oct 28, 2010 at 1:42 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 10/28/2010 12:52 PM, Gordon Henderson wrote:
On Thu, 28 Oct 2010, Jonas Kellens wrote
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone?
If
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