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mazilo
From Mazilo
Premium
join:2002-05-30
Lilburn, GA
| reply to cbrain
Re: Real reason why Grandcentral has 'press 1 to accept the call
said by cbrain  :Try Asterisk, you could have it automatically send the 1. Why not tell us how to do this on asterisk? | |
bellhater
Premium
join:2003-10-08
canada
| 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! | |
cbrain
join:2000-05-21
Silver Spring, MD
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| reply to mazilo
said by mazilo :said by cbrain :Try Asterisk, you could have it automatically send the 1. Why not tell us how to do this on asterisk? Because I've never done it. 
Sounds like you are using Trixbox. The above post looks like a good start. This would go in - extensions_custom.conf. If you use FreePBX to do your incoming config, you could start with "inbound routes" but send to "custom app" in "extensions_custom.conf" then from there send to an extension or ring group. If you use the fax handling remember your call is already answered. Below are a few commands you might play with. Let us know how you make out.
[extensions_custom.conf]
exten => s,n,Playtones(ring) ;play ringtone to caller if needed
exten => s,n,Goto(ext-group,1,1) ;go to ring group 1
exten => s,n,Goto(ext-local,1000,1) ; go to extension 1000 | |
hoolahoous
join:2004-08-25
Red Valley, AZ
| reply to bellhater
said by bellhater :here is an example of how I have done it in the past: thanks.. i will work on this and post freepbx/trixbox instructions if successful.. | |
tommy13v
Premium
join:2002-02-15
Glenville NY
1 edit | reply to bellhater
Very nice dialplan addition. Gotta love Asterisk and open source in general. | |
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