Hi,
I am also looking for a fax solution. I use Cisco ATA adapter with fax machine
and sending of fax works perfectly. But receiving fails after call is
answeredby the fax machine. It may be a problem of the fax machine also. The
configuration is simple. I use CISCO SPA 112 – 2 Port Phone
Ok, thanks for all
Best Regards
Marco
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Joshua Colp
Inviato: mercoledì 26 settembre 2012 19:37
A: Asterisk Users Mailing List - Non-Commercial Discussion
I have registered in sip.conf and in my network i am not using any port
forwarding kind of stuff (NAT), Asterisk server is directly connected to
Internet and the Internet router doesn't have any firewall.
And attached is asterisk log, that SIP REGISTER messages keep on sending
and no response
I have registered in sip.conf
wow that was very detailed. I think I asked How have you configured this to
register ? I'm pretty much sure you've nat related string mis-configured in
your sip.conf.
Can you tell if your asterisk server has access to internet !! I can see
the same situation
Am 27.09.2012 08:15, schrieb Shanavaz E A:
Patrick, can you please give the steps to configure fax with iaxmodem
and hylafax. Is it free to use?
I'm not Patrick, but here's a good how-to that worked perfectly for me.
In German, but it's basically enough just to type the commands :)
And yes,
What do you get if you run a queue show sales?
l.
2012/9/26 Mitch Claborn mitch...@claborn.net
Asterisk 1.8.10.1~dfsg-1ubuntu1
Trying to build a simple announcement of the queue status. QUEUEHOLDTIME
is always zero. What am I doing wrong?
queues.conf
[general]
autofill=yes
I'd go for MyISAM and would set up a remote replica if data integrity is
important.
If you have like 1000 calls of (say) 30 seconds avg length, and you create
10 events per call, you would expect an event every three seconds. This is
about 300 inserts per second. Say 600 at peaks. This should be
Hello,
this might seem a stupid question but I really don't see the solution to
the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
sipusers = mysql,AsteriskHosted,sip_buddies
sippeers = mysql,AsteriskHosted,sip_buddies
queues =
Maybe a stupid answer ;-)
Did you make a "reload"?
Did you try from shell:
mysql -u myuser -pmysecret AsteriskHosted
?
Am 27.09.2012 11:00, schrieb Jonas
Kellens:
Hello,
this might seem a stupid question but I
On 27-09-12 11:27, Thorsten Göllner wrote:
Maybe a stupid answer ;-)
Did you make a reload?
Yes, I reloaded and restarted several times.
Did you try from shell:
mysql -u myuser -pmysecret AsteriskHosted
Yes, works perfect to connect via commandline.
Only Asterisk does not see the
On Thu, 2012-09-27 at 11:00 +0200, Jonas Kellens wrote:
Hello,
this might seem a stupid question but I really don't see the solution
to the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
sipusers =
Now I see: you try to use the wrong config file - try
/etc/asterisk/res_config_mysql.conf instead.
Am 27.09.2012 11:40, schrieb Jonas
Kellens:
On 27-09-12 11:27, Thorsten Gllner wrote:
Maybe a stupid answer ;-)
On 27-09-12 11:54, Ishfaq Malik wrote:
On Thu, 2012-09-27 at 11:00 +0200, Jonas Kellens wrote:
Hello,
this might seem a stupid question but I really don't see the solution
to the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
Now I see: you try to use the wrong config file - try
/etc/asterisk/res_config_mysql.conf instead.
Am 27.09.2012 11:40, schrieb Jonas
Kellens:
On 27-09-12 11:27, Thorsten Gllner wrote:
Maybe a stupid answer ;-)
On 27-09-12 11:54, Ishfaq Malik wrote:
On Thu, 2012-09-27 at 11:00 +0200, Jonas Kellens wrote:
Hello,
this might seem a stupid question but I really don't see the solution
to the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
On Thu, Sep 27, 2012 at 2:39 AM, Mitch Claborn mitch...@claborn.net wrote:
Asterisk 1.8.10.1~dfsg-1ubuntu1
Trying to build a simple announcement of the queue status. QUEUEHOLDTIME
is always zero. What am I doing wrong?
queues.conf
[general]
autofill=yes
shared_lastcall=yes
yes this is the link http://www.callwithus.com/configuration am following,
and using the same, except type=friend i am using type=peer,
[general]
register = username:passw...@sip.callwithus.com
[callwithus]
type=peer
host=sip.callwithus.com
username=username
secret=password
qualify=no
-- Executing [812@LocalSets:1] NoOp(SIP/08000F3BE07C-0005,
queue status) in new stack
-- Executing [812@LocalSets:2] Set(SIP/08000F3BE07C-0005,
LOGGEDIN=1) in new stack
-- Executing [812@LocalSets:3] Set(SIP/08000F3BE07C-0005,
READY=0) in new stack
-- Executing
Satish I believe you have the answer. See output below, where I have 1
call answered and 1 in the queue. Unfortunately, the average wait time
is very inaccurate. These two calls where placed within seconds of each
other. The one still in the queue has a wait time of 4:10, so the
average
I am also writing an AMI application that will allow management to see
the queue status from an external program and saw the same issues with
the AMI data. Using AMI I am able to get what I need from the
individual records for each queued call.
Mitch
On 09/26/2012 04:09 PM, Mitch Claborn
We have customers that have migrated to our network from them due to their
reliability issues. Most of them are in the US west and east.
Jared Baxley jared.bax...@gmail.com wrote:
Has anyone had experience using a SIP trunk provided by Paetec over MPLS?
With or without FreePBX
Date: Thu, 27 Sep 2012 10:23:35 +0200
From: lenz.lo...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR
I'd go for MyISAM and would set up a remote replica if data integrity is
important.
If you have like 1000 calls of (say) 30 seconds
Did you try restarting asterisk not only a reload
Also I found a few broken stuff in queues like the rules (yes its on the
tracker) maybe this is also
-Original Message-
From: Mitch Claborn mitch...@claborn.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 27 Sep 2012
On Thu, Sep 27, 2012 at 9:15 AM, Mitch Claborn mitch...@claborn.net wrote:
Satish I believe you have the answer. See output below, where I have 1
call answered and 1 in the queue. Unfortunately, the average wait time is
very inaccurate. These two calls where placed within seconds of each
I did try restarting asterisk - no difference.
mitch
On 09/27/2012 10:59 AM, isr...@gmail.com wrote:
Did you try restarting asterisk not only a reload
Also I found a few broken stuff in queues like the rules (yes its on the
tracker) maybe this is also
-Original Message-
From:
Warren - that coincides with what I am seeing. I guess it made sense to
someone, but it is not terribly useful to me.
mitch
On 09/27/2012 11:22 AM, Warren Selby wrote:
On Thu, Sep 27, 2012 at 9:15 AM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Satish I
Dear Leif Madsen,
Please explain more
On Wed, Sep 26, 2012 at 11:31 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 26/09/12 05:35 AM, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music
The idea is a busy call center that takes calls all day long should be able to
determine their average wait / hold time over the course of the day. It's a
metrics thing, not a live data feed. It doesn't really become useful until
you've had several live calls, and then it's only useful if
On 09/27/2012 08:15 AM, Shanavaz E A wrote:
[snip]
Patrick, can you please give the steps to configure fax with iaxmodem
and hylafax. Is it free to use?
It's been years since I set it up so I don't know exactly how to
configure it anymore. But I do remember that I found some howto/docs via
Hi,
Is there a way to move 100 .call files in to
/var/spool/asterisk/outgoing/ at once and have Asterisk call at
maximum 10 at a time?
The maxcalls setting in asterisk.conf will limit the number of calls
but not gracefully. The calls over 10 just fail. I need to queue them
up.
I can certainly
Paetec was purchased by Windstream. I was looking one time ago to buy sip
trunk from them that run via T1. If anyone use them with asterisk it would
be nice to hear feedback.
On Sep 26, 2012 9:25 PM, Jared Baxley jared.bax...@gmail.com wrote:
Has anyone had experience using a SIP trunk provided
On 09/28/2012 03:01 AM, Patrick Archibald wrote:
Hi,
Is there a way to move 100 .call files in to
/var/spool/asterisk/outgoing/ at once and have Asterisk call at
maximum 10 at a time?
Afaik that is not possible. Wouldn't it make more sense to move call
files in batches of 10 to outgoing/?
I agree. A script that read the spool directory, sent enough files to equal
10, wait a few seconds, check again and move more would do the trick.
- Logan
On Sep 27, 2012 11:27 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl
wrote:
On 09/28/2012 03:01 AM, Patrick Archibald wrote:
Hi,
Is
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