bellhater
Premium
join:2003-10-08
canada
| reply to mazilo
I have done it.. Basicly in your inbound context here is an example of how I have done it in the past:
The following is an example of how to allow asterisk to take the call and voicemail.
exten => s,1,answer
exten => s,2,wait(2)
exten => s,3,SendDTMF(1)
exten => s,4,voicemail(U200)
exten => s,5,hangup
Now another example would be the same, however directed to your voip device, ring for 20 seconds and then go to voicemail if no answer from device1:
exten => s,1,answer
exten => s,2,wait(2)
exten => s,3,SendDTMF(1)
exten => s,4,dial(sip/device1,20)
exten => s,5,voicemail(u200)
exten => s,6,hangup
And lastly if you wanted to use asterisk to transfer the call to your home phone or any other phone number with the ability to allow voicemail on that phone number, if the line is busy, the line is busy in the context and will ring busy.. sometimes this confuses the caller, cause they will hear the gc ring, then a busy signal:
exten => s,1,answer
exten => s,2,wait(2)
exten => s,3,SendDTMF(1)
exten => s,4,dial(sip/17018255555@provider)
exten => s,5,busy
The information provided is for advanced users. If your a beginner I would offer my services to help for a very small fee as there are several interlinks to make this work as follows:
1:DID number from a provider
2:Asterisk server, virtual or physical (free open source ver)
3:Outbound provider (if allowing your calls to ring to a landline or cell phone)
You can use the gizmo project account to make the interconnect if you wanted to save some money and go virtually free from gc to voip device with asterisk passing the call.
Good luck! |
