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I need to configure an Asterisk box to go to voicemail but only after X ringouts.

exten => 1234,1,Dial(SIP/ivan, 30)
exten => 1234,2,VoiceMail(777@mb_tutorial)
exten => 1234,3,PlayBack(vm-goodbye)
exten => 1234,4,HangUp()

The client in my case is a softphone client. If it does not running or not connected to the pbx then this would immediately redirect the caller to the voicemail. The 30 seconds wait time does not apply in this case when the phone (sip client) is unavailable.

How could I configure Asterisk to try to ring out at least 5 times (regardless the sip client is offline) then if there is still no answer drop the caller to the voicemail.

Thanks

astlinux
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3 Answers3

2

You can create a context in extensions.conf that just rings forever if you call it, and then dial both the extension you want to reach and that contex. This guarantees that it will always run the Dial() application for 30 seconds before jumping to the voicemail. (Change the 30 seconds value as needed to get five ringtones... the period of ringtones can be found in the appropriate section of indications.conf for your country, so you can calculate how much time five rings are supposed to be.)

It can look something like this:

[noop]
exten => s,1,Wait(30)
same => n,Hangup

[your_context]
exten => 1234,1,Dial(SIP/ivan&Local/s@noop,30)
exten => 1234,2,VoiceMail(777@mb_tutorial)
exten => 1234,3,PlayBack(vm-goodbye)
exten => 1234,4,HangUp()

Calling the noop context will basically have no effect, and if SIP/ivan is available, then it will ring. If you are not hearing ringtones when calling the 1234 extensions, you can try different options:

  1. First possibility, using call progress (avoiding answering the channel and thus having the caller possibly pay to hear ringtones... but does not always work, depending on the SIP provider):

    [your_context]
exten => 1234,1,Progress
exten => 1234,2,Ringing
exten => 1234,3,Dial(SIP/ivan&Local/s@noop,30)
exten => 1234,4,VoiceMail(777@mb_tutorial)
exten => 1234,5,PlayBack(vm-goodbye)
exten => 1234,6,HangUp()

  2. Second possibility, generating ringtones as sound on the answered channel (meaning that the caller pays for the call while waiting for an answer too, because the call is actually answered):

    [your_context]
exten => 1234,1,Answer
exten => 1234,2,Playtones(ring)
exten => 1234,3,Dial(SIP/ivan&Local/s@noop,30)
exten => 1234,4,VoiceMail(777@mb_tutorial)
exten => 1234,5,PlayBack(vm-goodbye)
exten => 1234,6,HangUp()

    If you do not hear the ring tones while waiting, try adding a ,r parameter to the Dial application:

    exten => 1234,3,Dial(SIP/ivan&Local/s@noop,30,r)
Ale
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0

There are multiple choices based on how you're using Asterisk.

  1. Using FreePBX or a similar interface:

Setup a ring group and set it to "ringall". You can then specify how many seconds it will ring.

  1. Using hand written dialplan code:

Setup a ringall queue in queue.conf

[ivan]
strategy = ringall
context = default
member => SIP/ivan

Adjust your dialplan similar to this:

exten => 1234,1,Answer
exten => 1234,n,Queue(ivan,t,,,30)
exten => 1234,n,VoiceMail(777@mb_tutorial)
exten => 1234,n,PlayBack(vm-goodbye)
exten => 1234,n,HangUp()
dkwiebe
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0

The second answer cannot be considered as a solution, also the syntax changed long time ago.

It should be Queue(ivan,t,,,30) instead of Queue(ivan|t|||30).

In this case it does ring out to the SIP peers but those are "fake rings" the call is already established, therefore the caller have to pay the fee even for ringing. The first one I will try Today but I have a feeling that the problem with that will be that if Ivan is Offline and comes back online during the 30 sec period, he won't get the rings anymore. I hope I'm not right.

Thanks for the answers anyway.

snowlinux
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