Log into the Linksys GUI, Look at what SIP account it is registering to
asterisk with then run sip show users ?
- Original Message -
From: Hassan Abdalla hagga...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Saturday, 26 May, 2012 6:08:59 PM
Subject: [asterisk-users] Linksys PAPT2
Actually I understood that register line is not required, also since my PBX
is behind the pfsense firewall, now what i am going to do is putting the
PBX directly in public network (i.e. without firewall) and will check whats
going to happen.
Hope things would sort out.
Regards.
On Sat, May 26,
Hello List,
Someone please address my issue...
On Thu, May 24, 2012 at 2:08 PM, Ashish Agarwal ashisha...@gmail.comwrote:
Hello,
I am trying to achieve call record file and play the file to the user
using phpagi script.
I am using the following code
$rand=/var/sounds/145712900;
The wct4xxp module will log kernel messages when starting up, indicating
whether a VPM module (VPM400 or VPM450) was found.
I have some systems that have been running long enough that the messages
files from the last reboot have long since been rotated out and deleted.
They are older systems
On 05/28/2012 05:18 AM, Tony Mountifield wrote:
The wct4xxp module will log kernel messages when starting up, indicating
whether a VPM module (VPM400 or VPM450) was found.
I have some systems that have been running long enough that the messages
files from the last reboot have long since been
In article 4fc368d6.5060...@digium.com,
Kevin P. Fleming kpflem...@digium.com wrote:
On 05/28/2012 05:18 AM, Tony Mountifield wrote:
The wct4xxp module will log kernel messages when starting up, indicating
whether a VPM module (VPM400 or VPM450) was found.
I have some systems that have
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On 2012-05-25 16:47, jose carlos de souza wrote:
Hi,
I need help.
Sorry for my bad english, i'm from Brasil. And just starting to leran
Linux because of Asterisk.
I want to configure Asterisk 1.8 with ipv6 in my Lan network at home.
I've done a clean install of it and SIP clients it's
Administrator TOOTAI wrote:
we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn't accept our calls anymore, we receive a
timeout error Packet timed out after 32000ms with no response.
Hi,
I´m facing some issues on asterisk 1.8.10. I can see this on the console:
[May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock:
audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex:
Operation not permitted
[May 28 15:46:19] ERROR[28099]: lock.c:280
Hello all,
I quickly searched online prior to sending this, but if this was previously
brought up, I sincerely apologize.
I noticed IMAP Voicemail remains as a build option in later versions of
Asterisk 1.8 but the only latest version I've found working is 1.8.8.0.
Is this an error in
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