[asterisk-users] CALL TRANSFER

[Damon Estep]

Be careful, if you set both T and t you might be allowing the wrong
party to transfer the call! In MOST cases you would want T or t, not T
and t, although there are some cases where you might want both.

 

 

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CALL TRANSFER

 

Thanks!!!

 

I forget Tt option! (too basis!!)

 

On 12/1/06, Damon Estep <damon at suburbanbroadband.net> wrote: 

Your dial string must have either the t or T option set.

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com ] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Subject: [asterisk-users] CALL TRANSFER

 

Hi Guys,

 

I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working
fine. Rightnow I would like to enable Call tranfer (like Traditional
PBX) in Blind Transfer and AttendXFER, I'm reading features.conf in
accordance to voip-info.org <http://voip-info.org/>  but the transfer
doesn't work!  Please if you can provide me some examples will be very
appreciate. 

 

Rgds.

-- 
Omar E.P.T
-----------------
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp 
Open Source Solutions
www.usysnet.com.pe <http://www.usysnet.com.pe/>  


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-- 
Omar E.P.T
-----------------
Certified Networking Professionals make better Connections!

http://omarept.blogspot.com/

  Usysnet Corp
Open Source Solutions
www.usysnet.com.pe 

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