I made the changes you told me, but, when I try to make an incoming call, in the Asterisk console, I get....
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/9'
-- Executing Dial("SIP/aurelio-92fe", "IAX2/501050:[EMAIL PROTECTED]/501050|60|r") in new stack
-- Called 501050:[EMAIL PROTECTED]/501050
-- Call accepted by 65.39.205.121 (format ULAW)
-- Format for call is ULAW
-- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4
-- Executing Goto("IAX2/[EMAIL PROTECTED]:4569/14", "incoming|s|1") in new stack
-- Goto (incoming,s,1)
-- Executing Wait("IAX2/[EMAIL PROTECTED]:4569/14", "1") in new stack
-- Executing Answer("IAX2/[EMAIL PROTECTED]:4569/14", "") in new stack
-- Executing DigitTimeout("IAX2/[EMAIL PROTECTED]:4569/14", "10") in new stack
-- Set Digit Timeout to 10
-- Executing ResponseTimeout("IAX2/[EMAIL PROTECTED]:4569/14", "20") in new stack
-- Set Response Timeout to 20
-- Executing BackGround("IAX2/[EMAIL PROTECTED]:4569/14", "itranser/msg_bienvenida") in new stack
-- Playing 'itranser/msg_bienvenida' (language 'en')
-- IAX2/65.39.205.121:4569/13 answered SIP/aurelio-92fe
-- Channel 'IAX2/[EMAIL PROTECTED]:4569/14' unable to transfer
-- Hungup 'IAX2/65.39.205.121:4569/13'
Why I get an "Unable to transfer" error on this channel?
How could I solve this problem?
Any clue will be wellcome
Thanks a lot.
Ismael Gil.
Christopher Lee wrote:
I having a lot of troubles to configure a simple voice menu.
In extensions.conf I have the following.
[incoming]
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,3,DigitTimeout,10
exten => s,4,ResponseTimeout,20
exten => s,5,Background(itranser/msg_bienvenida)
exten => 1,1,Goto(contexto_extensiones,s,1)
exten => 2,1,Goto(contexto_operadora,s,1)
The context refered by the menu. (each context play me a
diferent message only )
[contexto_operadora]
exten => s,1,Background(itranser/trans_operadora)
exten => s,2,Dial(SIP/ismael,s,1)
[contexto_extensiones]
exten => s,1,Background(itranser/msg_pasar_ext)
I've made the corrections to your context's above... Note in particular
the Goto command and then using the 's' (start) extension in each
extension line, also adjusted the priority numbers.
For more info on Goto
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Goto
Give that a try and see how you go.
Regards,
Chris Lee
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