[asterisk-users] GXP-2020 Transfer Key
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with asterisk?. Attended and blind transfer does not work wiith this IP Phone Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and TDM400P
Hi, I have a problem with a TDM400P card configuration. Incoming calls are answered by asterisk, asterisk place the call on the destination ATA/analog-phone, the phone begins to ring and when our recepcionist pickup the phone to play a welcome message, she nothing hear on the line during five or six seconds repeating the message three or four times until someone appear on the phone. I was playing with Wait application but nothing changes this issue. On the other hand I disabled callerid from zapata.conf but this problem continues. I'm using Asterisk 1.2.24 on a debian system I hope have exposed the problem crearly. Thanks for any idea to solve this issue. Cheers Alejandro González ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Glare on Incoming Calls
How I change my configuration to reduce this issue. I have this setting on
my zapata.conf
signalling=fxs_ks
group=1
callgroup=1
pickupgroup=1
channel=1
signalling=fxs_ks
group=2
callgroup=1
pickupgroup=1
channel=2;
singalling=fxs_ks
group=3
callgroup=1
pickupgroup=1
channel=3;
singalling=fxs_ks
group=4
callgroup=1
pickupgroup=1
channel=4
and for outbound calls I have this context on my extensions.conf
[out-callb]
exten = 44,1,Set(LANGUAGE()=es)
exten = 44,n,ChanIsAvail(Zap/g1Zap/g2Zap/g3Zap/g4)
exten = 44,n,GotoIf($[${AVAILCHAN} = ]?4:6)
exten = 44,n,Congestion
exten = 44,n,Hangup
exten = 44,n,Playback,ggestion/varios/moment
exten = 44,n,SetMusicOnhold(dialtone)
exten = 44,n,Set(TIMEOUT(response)=10)
exten = 44,n,Set(TIMEOUT(digit)=5)
exten = 44,n,WaitExten(25|m(dialtone))
Date: Thu, 18 Oct 2007 17:07:03 -0400
From: C F [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Incoming calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
Glare that's what it's called, if the number you advertise as your
business number is zap/1 then use zap/G1 to dial out, otherwise use
zap/g1 to dial out. This will reduce but not eliminate the problem.
On 10/18/07, Gustavo Gonzalez [EMAIL PROTECTED] wrote:
Hello I have a question about incoming calls on TDM400P cards. I want to
know why an incoming call appear in a sorpresive way on a phone that I
pickup to call out. I am using ChanIsAvailable to check those lines ( Zap
channels )that are free. I have four lines connected to my TDM400P card
and
when I get a free Zap channel to call I hear the voice of a people on the
other side from an incomming call, I think that asterisk bridge my free
channel with incomming calls but how do this?Thanks for any idea.
Alejandro González
Grupo Gestión
4384-0660
www.grupo-gestion.com.ar
[EMAIL PROTECTED]
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[asterisk-users] Incoming calls
Hello I have a question about incoming calls on TDM400P cards. I want to know why an incoming call appear in a sorpresive way on a phone that I pickup to call out. I am using ChanIsAvailable to check those lines ( Zap channels )that are free. I have four lines connected to my TDM400P card and when I get a free Zap channel to call I hear the voice of a people on the other side from an incomming call, I think that asterisk bridge my free channel with incomming calls but how do this?Thanks for any idea. Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006 01:45 p.m. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P
Hello all, i've configured a TDM400P card but some calls hangs up and when i take the phone to do a call y hear someone that callme. How is the way to check the line before to do a call?. Other thing, is there a way to use Dial application without ring the phone if the line is busy or unavailable?. I want to get busy channel or unavailable channel whitout ring or music on hold if the phone are busy or no available. Thanks!! Alejandro González ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Line State
Hello all, I need a little help to check the state of the line from asterisk on aa TDM400P because when the telco lines goes down, asterisk get that line for outgoing calls. There is a way to check it out? And when all lines are busy to do outgoing calls how can i do to callback the people that call when a line is free? Thanks Alejandro González Grupo Gestión 4384-0660 www.grupo-gestion.com.ar [EMAIL PROTECTED] --- --- RI 9000-1069 Sistema de Gestión de Calidad Certificado por IRAM Norma ISO: 9001-2000 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: 18/12/2006 01:45 p.m. attachment: image001.gif___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P
Hi all! I have an issue with TDM400P FXO card. When a call enter into my IVR and select the proper option, the person that ansswer the call say your thanks for contact us ... but the caller cant hear this words because a delay between asterisk and caller part or between asterisk and the ATA device. What is the item on zapata.conf that can affect this delays. Thanks for any help Alejandro González ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
