Re: [asterisk-users] redirect a ringing phone

2011-12-09 Thread Jerry Geis

 On 12/07/2011 10:16 AM, Jerry Geis wrote:
 I am using AMI to call a phone and play a wave file. That works fine 
to SIP/401.


Now I am trying to redirect that call that is ringing to another 
phone (SIP/404).
When I do it the other phone rings but the first phone continues to 
ring also.


Then when I answer on SIP/404, I get a ring not the wave file.

 Action: Setvar
 Channel: SIP/401-0004
 Variable: SMVOICE_CALLAT
 Value: SIP/404
DEBUG: Response: Success[CR ][LF ]Message: Variable Set[CR ][LF ][CR 
][LF ]


Action: Redirect
Channel: SIP/401-0004
Exten: smvoice_dial_no_extension
Context: smvoice-transfers
Priority: 1
DEBUG: Response: Success[CR ][LF ]Message: Variable Set[CR ][LF ][CR 
][LF ]


exten = 
smvoice_dial_no_extension,1,Dial(${SMVOICE_CALLAT},${SMVOICE_DIAL_TIMEOUT},${SMVOICE_ONHOLD}tT)



What am I not issuing correctly to redirect the ringing phone to 
another phone.


Thanks,

jerry

I forgot to include that I was using asterisk 1.4.42.
I had not seen any replies . Is there information I am missing?
Am I doing something wrong?

jerry

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[asterisk-users] redirect a ringing phone

2011-12-07 Thread Jerry Geis
 I am using AMI to call a phone and play a wave file. That works fine 
to SIP/401.


Now I am trying to redirect that call that is ringing to another phone 
(SIP/404).
When I do it the other phone rings but the first phone continues to ring 
also.


Then when I answer on SIP/404, I get a ring not the wave file.

 Action: Setvar
 Channel: SIP/401-0004
 Variable: SMVOICE_CALLAT
 Value: SIP/404
DEBUG: Response: Success[CR ][LF ]Message: Variable Set[CR ][LF ][CR ][LF ]

Action: Redirect
Channel: SIP/401-0004
Exten: smvoice_dial_no_extension
Context: smvoice-transfers
Priority: 1
DEBUG: Response: Success[CR ][LF ]Message: Variable Set[CR ][LF ][CR ][LF ]

exten = 
smvoice_dial_no_extension,1,Dial(${SMVOICE_CALLAT},${SMVOICE_DIAL_TIMEOUT},${SMVOICE_ONHOLD}tT)



What am I not issuing correctly to redirect the ringing phone to another 
phone.


Thanks,

jerry

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users