I need help need recording all call for my pbx but i am a novato in asterisk
my confi for record is:
exten=_NX,n,Set(CALLFILENAME=CLIENTE-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${UNIQUEID})
exten = _NX,n,MixMonitor(${CALLFILENAME}.gsm,m)
exten =
i need a program for monitorin my bandwitch of my asterisk server
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Bayardo Sánchez García
Web Developer - Internet Portals
Linux User: #418392
Ubuntu User #14171
America Central - Managua, NI (505) 249-2853 - 4886876
IM msn messenger: bjsanch...@hotmail.com
Skype: bayardo.sanchez
This email
I have a problem when I call a good record but I make a call to return to
the same number I erased the previous record, and I replaced with the last
call
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Bayardo Sánchez García
Web Developer - Internet Portals
Linux User: #418392
Ubuntu User #14171
America Central - Managua, NI (505)
i am used eyebeam is a good softphone
On Sun, Jan 25, 2009 at 12:55 PM, David fire ddf...@gmail.com wrote:
hi
wich soft phone do you recomend but i need this feature it must ask for
user name and password when it start.
i know xline and zoipper but they dont have that i can acomplish this
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
'SIP/8022-b7225740'
-- Got SIP response 503 Service Unavailable back from 74.63.41.218
-- SIP/voipms4-09ab0c38 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto
The problem is I have 20 agents calling all but 3 of them get this error
when calling
On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote:
Bayardo Sanchez wrote:
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
'SIP/8022
24 chanels
On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com wrote:
What is your call-limit set to in sip.conf?
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
only 3
On Tue, Jan 27, 2009 at 10:03 AM, Danny Nicholas da...@debsinc.com wrote:
Is it the same 3 or the first 3?
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
*Sent
i any problem conecct ready
On Sat, Jan 31, 2009 at 7:37 AM, Jonn Taylor jo...@taylortelephone.comwrote:
Anyone else having problems connecting to
http://downloads.digium.com/pub/ ??
Jonn
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i create inbound number but i calling and send this error:
[Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension '246463' rejected because
extension not found.
but the extensin existed
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Bayardo Sánchez García
Web Developer -
in my extension.conf i set :
[default]
exten = 1246463,1,Answer(SIP/8003)
On Mon, Mar 16, 2009 at 12:06 PM, Pascal Bruno tipas...@gmail.com wrote:
Do you have an extension set for 246463 in your extensions.conf?
On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez
bayardo.sanc
nothing the problem persitem
On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote:
are you sure calls from this provider are going to context 'default' ?
sip.conf
[procall]
type=peer
username=XX
secret=XX
context=default
2009/3/16 Bayardo Sanchez bayardo.sanc
...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
*Sent:* Monday, March 16, 2009 4:38 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Help Inbound number
nothing the problem persitem
On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com
i create inbound confi my confi is:
[incoming]
exten= 1246463,,1,Dial(SIP/8003,60,rT)
exten= 6463,1,Dial(SIP/8003,60,rT)
exten= 1246463,,n,Wait(5)
exten= 1246463,,n,Hangup
but y calling and send this error in my CLI:
[Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383
, Bayardo Sanchez wrote:
i create inbound confi my confi is:
[incoming]
exten= 1246463,,1,Dial(SIP/8003,60,rT)
exten= 6463,1,Dial(SIP/8003,60,rT)
exten= 1246463,,n,Wait(5)
exten= 1246463,,n,Hangup
but y calling and send this error in my CLI:
[Apr 15 09:58:48] NOTICE
i call my tollfree number and send the call to my extension 8003
On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. bran...@brellsystems.comwrote:
You call call to extension '246463' will not match 'exten =
1246463'.
On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez
bayardo.sanc
,Dial(SIP/8003,60,rT)
exten= 246463,n,Wait(5)
exten= 246463,n,Hangup
exten= 6463,1,Dial(SIP/8003,60,rT)
On Wed, Apr 15, 2009 at 11:00 AM, Bayardo Sanchez
bayardo.sanc...@gmail.com wrote:
i call my tollfree number and send the call to my extension 8003
On Wed, Apr 15, 2009 at 10:51
tollfree calls was working fine but stopped working without any reason
On Wed, Apr 15, 2009 at 11:51 AM, Brandon B. bran...@brellsystems.comwrote:
Is your system configured to send the dialed calls to the [incoming]
context?
On Wed, Apr 15, 2009 at 11:22 AM, Bayardo Sanchez
bayardo.sanc
i call to my tollfree number buy my CLI send the next error:
Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected
freqency 22050
Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open
file on /var/lib/asterisk/sounds/procall3.wav
Aug 14 08:15:22
Of *Bayardo Sanchez
*Sent:* Friday, August 14, 2009 9:59 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] i have a error in ivr
i call to my tollfree number buy my CLI send the next error:
Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header
exten = t,1,goto(procall,s,1)
exten = h,1,Hangup
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
*Sent:* Friday, August 14, 2009 10:20 AM
*To:* Asterisk Users Mailing List - Non
are using some kind of router?
On Sat, Sep 26, 2009 at 8:20 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello
I Recently completed an IVR application with Asterisk.
Now we are moving towards VOIP. Please give a direction how to move
forward.
What i have studied so far
I am
I have a user but I need to give that user only kill and disable all
connection cut calls what is the command in the CLIC
--
Bayardo Sánchez García
Web Developer - Internet Portals - Asterisk Support - Windows Server Support
- Proxy Support - Linux Server
E-mail: bayardo.sanc...@gmail.com
Linux
I'm working in callcenter I using vicidial is cool
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Sent from my BlackBerry®
VoIP, Windows/Linux Administration and Network Management
-Original Message-
From: wassim darwich wassimdarwi...@yahoo.com
Date: Sun, 7 Mar 2010 06:21:34
To: asterisk-users@lists.digium.com
Subject:
This week I was experiencing attacks sip log into my accounts were more than
1,000 requests for records Sip accounts in less than an hour THROUGH deny the
ip of my router access list in cisco and my asterisk server to go through the
iptables drop ip attacker is a way for an account with
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x85a6420', 9
24.11.2010 13:48, schrieb Bayardo Sanchez:
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780
, schrieb Bayardo Sanchez:
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
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Sent from my BlackBerry®
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Sent from my BlackBerry®
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Sent from my BlackBerry®
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Download DriveCarefully for free at www.drivecarefully.com
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