Re: [asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})

2010-02-18 Thread Mariano Lecuona
...@selbytech.com On Wed, Feb 17, 2010 at 2:47 PM, Mariano Lecuona mlecu...@gmail.comwrote: Could anyone get the MONITOR_FILENAME set from the queue.conf with variables like: MEMBERINTERFACE is the interface name (eg. Agent/1234) MEMBERNAME is the member name (eg. Joe Soap) MEMBERCALLS is the number

[asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})

2010-02-17 Thread Mariano Lecuona
All, I am trying to set a monitor file from the queue.conf as specified on http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf In order to avoid the default MONITOR_FILENAME format wich is: agent-x-uniqueid.wav for example agent-10017-1266438575-26.wav As you may now, when using

Re: [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)

2010-02-12 Thread Mariano Lecuona
As far as my experience, this problem occurs when the asterisk tries to take a new channel and teco does not count with any available channels. Contact your E1/T1 provider and work with them to search on the teco side. 2010/2/12 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Feb 12, 2010 at

Re: [asterisk-users] Can an agent Login to a queue and be paused

2010-02-08 Thread Mariano Lecuona
What Id did was on the dialplan, create an specifica extension for login agents. Lets say Agent/10017, then When dial 2110017 the agents is promts for Agent passwd.Then I have a macro only for pausing agents depending on the meaning. So if the agent is successfully granted on the Login Context,

Re: [asterisk-users] Can an agent Login to a queue and be paused

2010-02-08 Thread Mariano Lecuona
... -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mariano Lecuona *Sent:* Monday, February 08, 2010 8:20 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Can an agent Login to a queue

[asterisk-users] NVFaxDetect

2010-02-01 Thread Mariano Lecuona
Hi all, Do anyone has a detailed procedure for NV_application install? I have search as I was told, but I did no find any thing accurate. Thanks ML -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

[asterisk-users] FAX over ISDN PRI

2010-01-30 Thread Mariano Lecuona
Hi, All I just want to be able to detect the fax signail while doing an outbout call taking advance of the out_dialout feature of asterisk. So for to have a clear image on how i am doing it. I have my .call that I move the /var/spool/asterisk/ouotgoing like: (numbers were changed to preserve

Re: [asterisk-users] FAX over ISDN PRI

2010-01-30 Thread Mariano Lecuona
thanks.. 2010/1/30 Kevin P. Fleming kpflem...@digium.com Mariano Lecuona wrote: All I just want to be able to detect the fax signail while doing an outbout call taking advance of the out_dialout feature of asterisk. So for to have a clear image on how i am doing it. The faxdetect

Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

2009-11-03 Thread Mariano Lecuona
of the major advantages of ACBL is that the set of queues is statically defined, so at the dialplan level you do not have to keep track of the set of queues an agent is enabled on. l. 2009/11/3 Mariano Lecuona mlecu...@gmail.com My mental plan orginilly was: 1.- Creating a macro that acceps

Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

2009-11-02 Thread Mariano Lecuona
with older asterisk's - authentication using Voicemail -.plug and play on most systems - channel states - pause/unpause with pause codes - ...you tell me Anybody interested? l. 2009/10/30 Mariano Lecuona mlecu...@gmail.com Hi all, I would like to know if there is any application

[asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6

2009-10-30 Thread Mariano Lecuona
Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a

Re: [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic

2009-10-30 Thread Mariano Lecuona
Take a look at this document. This may help you on trouble shoot your kernel panic. http://www.novavox.co.uk/docs/install-guides/novavox-asterisk-card-installation-issues.pdf 2009/10/30 David Shauger sollost...@gmail.com Setting up a new Asterisk server with Centos 5.2 and Asterisk 1.4.23

Re: [asterisk-users] DAHDI not detecting RINGING Status on the Channel

2009-10-27 Thread Mariano Lecuona
FXSKS (In use) RED 6 FXOFXSKS (In use) RED 7 FXOFXSKS (In use) RED 8 FXOFXSKS (In use) RED 2009/10/27 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Oct 26, 2009 at 09:02:10PM -0300, Mariano Lecuona wrote: For some reason I am

[asterisk-users] DAHDI not detecting RINGING Status on the Channel

2009-10-26 Thread Mariano Lecuona
I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts,

Re: [asterisk-users] DAHDI not detecting RINGING Status on the Channel

2009-10-26 Thread Mariano Lecuona
in some instances (possibly applicable to you since you are non-US). -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mariano Lecuona *Sent:* Monday, October 26, 2009 2:59 PM *To:* asterisk-users

[asterisk-users] SIP Trunk groups

2009-05-27 Thread Mariano Lecuona
= sip_trunk1 ; sip_trunk2 ; sip_trunk3 [user] exten = _0.,1,wait() exten = _0.,n,Dial(gsm_trunkgoup/${ exten:1},30) exten = _0.,n,Hangup Thanks, -- -- *Mariano Lecuona* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

[asterisk-users] SIP Trunk groups

2009-05-25 Thread Mariano Lecuona
= sip_trunk1 ; sip_trunk2 ; sip_trunk3 [user] exten = _0.,1,wait() exten = _0.,n,Dial(gsm_trunkgoup/${exten:1},30) exten = _0.,n,Hangup Thanks, -- -- Mariano Lecuona ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk