[Asterisk-Users] Cisco ATA186 SIP transfer

[Jan Baumann]

Hello asterisk experts,

I have a running installation with a Cisco 7960 and an ATA186.
Attended and unattended transfer of an incoming PSTN call from 7960 to ATA works 
as expected.

 From ATA to 7960 users can press the flash button, dial the 7960, talk to the 
other ext. and should then be able to complete the transfer by hanging up 
according to Cisco's docs.

Instead, the connection is dropped at 7960 when ATA hangs up and the external 
call rings at the 7960 like a new call. So basically transferring works, but 
always requires hanging up in the middle.

Any ideas how to fix that?

Thank you and regards,

Jan Baumann


[did-from-pstn]
exten => 1234531,1,SetVar(ALERT_INFO=1)
exten => 1234531,2,LookupCIDName
exten => 1234531,3,Dial(SIP/31,20,t)
exten => 1234531,4,Voicemail2(u31)
exten => 1234531,5,Hangup
exten => 1234531,104,Busy

exten => 1234532,1,SetVar(ALERT_INFO=1)
exten => 1234532,2,LookupCIDName
exten => 1234532,3,Dial(SIP/32,20,t)
exten => 1234532,4,Voicemail2(u32)
exten => 1234532,5,Hangup
exten => 1234532,104,Busy

[from-sip-internal]
exten => 31,1,Dial(SIP/31,30,tr)
exten => 31,2,Voicemail2(u31)
exten => 31,3,Hangup
exten => 31,102,Busy

exten => 32,1,Dial(SIP/32,30,tr)
exten => 32,2,Voicemail2(u32)
exten => 32,3,Hangup
exten => 32,102,Busy