-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Gary Carr
Sent: Wednesday, October 03, 2012 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] call extension play
I am sorry, all my fault ): I used click2dial and set a wrong outgoing line
- snom lines count there lines starting at 1 instead of 0. how could they
only dare to do so...
2012/10/3 Tim Nelson tnel...@rockbochs.com
- Original Message -
No idea? ):
How about showing your dialplan,
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming and
outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful
of reasons.
Is it natural to have this many problems on a
Hi,
Can any one tell me on which linux kernel version i can compile and run the
DAHDI-2.0 release and test it .
*Regards
Upendra.*
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
I had the same problem for a while. I found replacing fax machines with a
scanner and either an email-to-fax program or just web-based faxing had
better results. I don't want to tell you the gateway I used because they
turned out pretty badly in the end. But there is hope!
- Logan
On Oct 4, 2012
Brett Lehrer wrote:
Hola,
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming and
outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful
of reasons.
I've never heard of
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten = s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten = s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME} = sip username from
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, October 04, 2012 9:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AVAILSTATUS always 0
Hello,
I notice that
On 04-10-12 16:59, Danny Nicholas wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, October 04, 2012 9:48 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:*
On 10/04/2012 09:29 PM, Brett Lehrer wrote:
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming and
outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful
of reasons.
Is
On Thu, Oct 04, 2012 at 07:40:57PM +0530, upendra wrote:
Hi,
Can any one tell me on which linux kernel version i can compile and run the
DAHDI-2.0 release and test it .
Probably 2.6.18 is your best bet. I'm genuinly curious as to why you would want
to run a version of dahdi from four years
Hi,
1. I've got this question bouncing in my mind for a long time: why are
alarm transmitters often said to be avoided with DSL lines ?
The kind of alarm transmitters I'm thinking about are those having two
analog ports: one connected to Telco analog line, the other to a fax or a
terminal or
2012/10/4 Brett Lehrer brett.leh...@solarismed.com
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming
and outgoing faxes, I'm getting a failure rate of just over 25%, and over a
handful of
On Thu, Oct 4, 2012 at 6:29 AM, Brett Lehrer brett.leh...@solarismed.comwrote:
I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking
service over a DSL line solely dedicated to VoIP usage. For both incoming
and outgoing faxes, I'm getting a failure rate of just over 25%,
On 10/04/2012 09:27 AM, Carlos Alvarez wrote:
However I'd just suggest that you look at the business case for
screwing around with fax at all. As a society, if we had decided to
stop supporting this dead technology years ago, with all the time and
money we've collectively wasted we could have
What is the setup you're talking about ?
Is it something like this ?
PSTN nexVortex T.38 gateway - Internet - DSL modem ---
Asterisk Fax machine
Olivier,
Sorry, I did a poor job explaining that. That's basically correct, with the
receiving end first and our originating end
On Thu, Oct 4, 2012 at 10:06 AM, Lee Howard fax...@howardsilvan.com wrote:
I recognize that you're being a bit facetious in this latter comment
No, not really. I stand by it. Useless and *should* be dead. It's dead
and people just don't know it.
There is no adequate replacement for fax.
I am trying to setup a context to take a inbound call, hold the
call,
connect to
an external number, play a sound file to the external number, then
connect
the inbound caller to the external number.
My thought was to accept the call and place them in a parking lot.
Then
call
On Fri, 28 Sep 2012 11:03:05 +0200
Jonas Kellens jonas.kell...@telenet.be wrote:
On 28-09-12 10:57, Administrator TOOTAI wrote:
Le 28/09/2012 10:40, Jonas Kellens a écrit :
Maybe I need to explain a bit further : the call is send to the
IP-phone and answered. The call lasts for about 1 à
On 04-10-12 19:50, Chad Wallace wrote:
On Fri, 28 Sep 2012 11:03:05 +0200
Jonas Kellens jonas.kell...@telenet.be wrote:
On 28-09-12 10:57, Administrator TOOTAI wrote:
Le 28/09/2012 10:40, Jonas Kellens a écrit :
Maybe I need to explain a bit further : the call is send to the
IP-phone and
Here is my IP-PBX setupmy setup is : sips softphone - asterisk - xorcom
PSTN gateway - pstn line to telcoi'm using xlite for windows
when I make a phone call (sip - outgoing channel),I can hear my own voice so
clear. it's very annoying mewhen talking a little loud... any solution?
Two
On Fri, Sep 28, 2012 at 7:42 PM, Mitch Claborn mitch...@claborn.net wrote:
I want to put a call me now button on the web site that will place the
request into an asterisk call queue and then when an agent picks up the
call in the queue, place the outbound call to the customer.
The following
From: Carlos Alvarez car...@televolve.com
Sent: Thursday, October 04, 2012 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fax for Asterisk success rates?
On Thu, Oct 4,
Hello:
I am investigating the possibility of using LDAP for storing certain Asterisk
configuration parameters.
I have examined res_ldap.conf and see where mailbox can be defined from
AstAccountMailbox but I do not see where the password can be stored ?
Am I missing something please ?
Thank
2012/10/4 Brett Lehrer brett.leh...@solarismed.com
What is the setup you're talking about ?
Is it something like this ?
PSTN nexVortex T.38 gateway - Internet - DSL modem ---
Asterisk Fax machine
Olivier,
Sorry, I did a poor job explaining that. That's basically correct,
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1(${thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
^
[Oct 4 21:53:35]
Hello All,
I am trying to debug an odd issue. I have two UACs that are sending
INVITEs to my asterisk 1.8 server. I want to start authenticating
these incoming invite requests with digest auth. The UACs are not
registered and I am using host ip to match them with a sip.conf peer.
The issue I
At 07:02 PM 10/4/2012, you wrote:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1(${thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
Try :
same=n,GoSubIf($[${CALLERID(num)}
On Thursday 04 Oct 2012, frangky robert wrote:
Here is my IP-PBX setupmy setup is : sips softphone - asterisk -
xorcom PSTN gateway - pstn line to telcoi'm using xlite for
windows when I make a phone call (sip - outgoing channel),I can hear
my own voice so clear. it's very annoying mewhen
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