On Wednesday 22 Aug 2012, Roberto Piola wrote:
> I would simply send the message with sendmail -v and then grep the
> output for the error message
Er, that works too :) Much better solution (as long as you are root).
Regards,
-- Raj
--
Raj Mathur || r...@kandalaya.org
Hi,
Can anyone tell me how to do the load test for the FXS and FXO cards and
find how much the asterisk machine can load for different processors
configuration .
Regards
Upendra.
--
_
-- Bandwidth and Colocation Provided by ht
I would simply send the message with sendmail -v and then grep the output
for the error message
Il giorno 22/ago/2012 04:19, "Raj Mathur (राज माथुर)"
ha scritto:
> On Tuesday 21 Aug 2012, Ruben Rögels wrote:
> > Hello,
> >
> > no problem at all, I think this is the tricky part.
> >
> > A smtp dia
On Tuesday 21 Aug 2012, Ruben Rögels wrote:
> Hello,
>
> no problem at all, I think this is the tricky part.
>
> A smtp dialogue between your email client and a smtp server normally
> looks like this:
>
> user@box:~? netcat mx1.example.com
> 220 postfix ESMTP mx1.example.com
> helo me.local
> 25
ot;SIP/1010-0162",
"MASTER_CHANNEL(REC_STATUS)=RECORDING") in new stack
-- Executing [s@macro-one-touch-record:9] Set("SIP/1010-0162",
"AUDIOHOOK_INHERIT(MixMonitor)=yes") in new stack
-- Executing [s@macro-one-touch-record:10] MixMonitor("SIP/1010-0
> I have an issue that I have been bumping up against. We have some
> inbound fax services and occasionally an inbound fax that
> successfully came in would fail to store it's references in the
> database.
>
> We are using a function in func_odbc to update a database table. We
> call the function
Hey all
I have an issue that I have been bumping up against. We have some inbound
fax services and occasionally an inbound fax that successfully came in
would fail to store it's references in the database.
We are using a function in func_odbc to update a database table. We call
the function f
She's talking about asterisk 11 not asterisk 1.8.11
-Original Message-
From: Phil Frost
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 21 Aug 2012 15:19:31
To:
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk
Up?
2012/8/20 Luis H. Forchesatto
> Thanks for your answer.
>
> The logs where posted at pastebin, here the links:
>
> - Working Phone: http://pastebin.com/q3pHcwna
> - Not working phone: http://pastebin.com/iiCHPMmn
>
>
> 2012/8/20 Rusty Newton
>
>> On 8/20/2012 7:19 AM, Luis H. Forchesatto wr
- Original Message -
>
>
>
>
> I’m trying to get my Asterisk 11 test box set up with XMPP, having
> troubles with JabberSend().
Hola!
The underlying issue here (finger didn't hit shift to turn 2 into @) has been
fixed in Asterisk 11 as of revision 371518. You can grab that specific f
I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles
with JabberSend().
My jabber.conf file is as follows:
[general]
debug=no
autoprune=no
[testaccount]
type=client
serverhost=my.jabber.server
username=myaccount@my.jabber.server
secret=mypassword
port=jabberport
usetls=yes
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost
Sent: Tuesday, August 21, 2012 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 11 - BLF on Custom devices
On 08/21/2012 02:52 PM, Noah Engelbe
On 08/21/2012 02:52 PM, Noah Engelberth wrote:
The short of the output is -- there is no console output showing ==
Extension Changed 302[hints] new state on the Ringing or InUse&Ringing
events -- only on InUse or Idle events (which matches what I'm seeing
on the phones).
Weird. I just did a t
> I'd do a packet capture -- ideally from the phone, or using your switch to
> mirror the phone's port -- and look for a SIP NOTIFY. Then
> we can know if a NOTIFY is not being sent, or if it's just not being
> processed as desired by your Cisco SPA 509G. If it's not there, do
> the same on you
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Richard Mudgett
> Sent: Monday, August 20, 2012 3:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asteris
On 21 Aug 2012, at 19:32, Ruben Rögels wrote:
> Hello,
>
> no problem at all, I think this is the tricky part.
>
> A smtp dialogue between your email client and a smtp server normally looks
> like this:
>
> user@box:~? netcat mx1.example.com
> 220 postfix ESMTP mx1.example.com
> helo me.local
Hello,
no problem at all, I think this is the tricky part.
A smtp dialogue between your email client and a smtp server normally
looks like this:
user@box:~? netcat mx1.example.com
220 postfix ESMTP mx1.example.com
helo me.local
250 mx1.example.com
mail from:
250 2.1.0 Ok
rcpt to:
450 5.7.1
On 08/17/2012 03:21 PM, Jerry Geis wrote:
On 08/17/2012 06:36 AM, Jerry Geis wrote:
On 08/13/2012 04:58 PM, Jerry Geis wrote:
On 08/13/2012 01:13 PM, Jerry Geis wrote:
I am getting a "beep beep beep" (like a busy or hangup sound) when
I am using my
AGI to start up a conf. (did not happen with
I'm sorry, how i can to check for the return code of the smtp session?
I've never done :p
Thanks,
Danilo
Il 21/08/12 19:05, Ruben Rögels ha scritto:
Okay, so have a look at "mailcmd=" option in voicemail.conf
"mailbox" will mean a "e-mail-box" in the next lines.
What you need to do is wirting
Okay, so have a look at "mailcmd=" option in voicemail.conf
"mailbox" will mean a "e-mail-box" in the next lines.
What you need to do is wirting a shell script or what ever to check for
the return code of the smtp session (normally it should be a 450 in case
of full mailbox).
In case of "450 m
I'll explain. I have an email account, danilo.dionisi @ outlook.it, with
a maximum size of 100MB. For example, my inbox is full, and Paris Hilton
( =P ) leaves me a voicemail message. I have to check the space of my
inbox, this space is completely full, so I do not have to delete the
voicemail
Hello
Check voicemail.conf
maxmsg = 100
And change it.
On Tue, Aug 21, 2012 at 12:52 PM, Danilo Dionisi
wrote:
> I'm sorry, I haven't been clear.
> I do not have to check the inbox on Asterisk, but I have to check the free
> space on a particular mailbox of Exchange software.
> It's possibl
The message must be deleted if sent to the recipient, otherwise it must
remain on the Asterisk machine when the recipient's mailbox is full.
Il 21/08/12 18:52, Ruben Rögels ha scritto:
just another thought: if you send the message by mail, do you need to
save it?
regards,
Ruben
--
--
I'm sorry, I haven't been clear.
I do not have to check the inbox on Asterisk, but I have to check the
free space on a particular mailbox of Exchange software.
It's possible with the pair Asterisk-Sendmail?
Il 21/08/12 18:45, Danny Nicholas ha scritto:
Assuming that you are using the standard
just another thought: if you send the message by mail, do you need to
save it?
regards,
Ruben
Am 21.08.2012 18:45, schrieb Danny Nicholas:
Assuming that you are using the standard 100 message limit, just check for
INBOX/MSG0100.txt and send the message.
-Original Message-
From: aster
Assuming that you are using the standard 100 message limit, just check for
INBOX/MSG0100.txt and send the message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi
Sent: Tuesday, August 21, 2012 11
Hi all,
I have a problem with voicemail. My boss has asked me to send via email,
the message that a user leaves on the voicemail. This is very easy. :)
After, he asked me to check before sending the email, if the receiver's
mailbox is full. If the mailbox is full, Asterisk should call the
recev
- Original Message -
> From: "CB"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, August 21, 2012 4:39:32 AM
> Subject: Re: [asterisk-users] alwaysauthreject=yes not working as expected
>
> > > Asterisk 1.4.42
First, even if you were right and you d
On Mon, Aug 20, 2012 at 8:20 PM, mailsvb wrote:
> Hi,
> you need to build Asterisk with SRTP support...
>
> wget http://sourceforge.net/projects/srtp/files/latest/download -O
> srtp-latest.tgz
> tar -zxvf srtp-latest.tgz
> ./configure --prefix=/libsrtp
> make && make install
>
> And for Asterisk..
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, August 21, 2012 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Which card to get?
O
>> We are investigating the possibility of using Asterisk in a KVM based
>> virtual machine
We have found that Asterisk in a VM with a digital card does not work (At least
under VMWare ESXi 5 with PCI pass though). The timing on the card tested with
dahdi_test were awful, sub 80% on average.
On Tue, Aug 21, 2012 at 09:33:39AM -0400, James B. Byrne wrote:
> We are investigating the possibility of using Asterisk in a KVM based
> virtual machine to handle connections to and from our HylaFax service.
> Our current set up uses a dedicated host with external fax modems.
> What I wish to kn
We are investigating the possibility of using Asterisk in a KVM based
virtual machine to handle connections to and from our HylaFax service.
Our current set up uses a dedicated host with external fax modems.
What I wish to know is what interface card would the list members
recommend for a proof o
On 12-08-21 07:04 AM, neo nortan wrote:
dear guys
plz tell me which version of asterisk is compatible with centos 5.7
(2.6.18-308.8.2.el5).
and which is the latest version.
Almost any version of asterisk should compile on any recent version of
CentOS.
You can find the latest versions of A
dear guys
plz tell me which version of asterisk is compatible with centos 5.7
(2.6.18-308.8.2.el5).
and which is the latest version.
regards
neo
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to A
> > Asterisk 1.4.42
> >
> > Set alwaysauthreject=yes in [general] section of sip.conf.
> > Restarted asterisk
> >
> > However when I attempt to register I still get:
> > [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from
> > '' failed for '121.98.1.1' -
> Wrong
> > password
> > [2012
Le 20/08/2012 17:02, Daniel Pocock a écrit :
On 20/08/12 16:23, Administrator TOOTAI wrote:
Hi,
I have to connect 3 asterisk servers,each of them being TLS server for
his clients and connected in both way in TLS with both others asterisk,
each having hi own Common Name. Is this possible?
I set
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