Outgoing calls goes to Temporarily Unavailable

Asterisk Asterisk Dialplan
1

Good day All,

I tried looking for an active post about my current issue but am not finding anything as yet.

I am using Asterisk 13.1.0

When trying to make an outgoing call and the users Cellphone goes to Voicemail it send back a busy message to asterisk. Instead of playing the voicemail prompts for the extension trying to call the number it ends the call with temporarily unavailable.

See cli output below
== Using SIP RTP CoS mark 5
– Executing [082490XXXX@Outgoing:1] NoOp(“SIP/202-000006f5”, “Internal Note”) in new stack
– Executing [082490XXXX@Outgoing:2] Dial(“SIP/202-000006f5”, “SIP/2712736XXXX/082490XXXX”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/2712736XXXX/082490XXXX
– SIP/2712736XXXX-000006f6 is making progress passing it to SIP/202-000006f5
– SIP/2712736XXXX-000006f6 redirecting info has changed, passing it to SIP/202-000006f5
– SIP/2712736XXXX-000006f6 is busy
== Everyone is busy/congested at this time (1:1/0/0)

And then the phone gives and error Temporarily Unavailable.

Any ideas on what may cause this? All other calls go through correctly. It is only when a call goes to voicemail.

Thank you in advance.

2

You have no code, in the dialplan, after the Dial, to direct the call to the voicemail application. In fact, you seem to have no code after the Dial, at all.

3

Thanks for the reply.

I am not talking about the voicemail function in Asterisk.

When I call 082 XXX XXX from an extension and that persons cellphone goes to voicemail it ends the call instead of the party that initiated the call hears the "You have reached the voicemail of "

This is my dialplan (I have changed it to allow everything to test and it still does the same)

exten => _X.,1,NoOp(Client Name - Internal call from Extension {CALLERID(num)} to Extension {EXTEN})
same => n,Dial(SIP/2712736XXXX/${EXTEN})
same => n,Hangup()

4

The downstream peer has ended the call. There is absolutely nothing that Asterisk can do to override that.

5

Downstream = SIP extension ?

6

There is no such thing as a SIP extension.

Downstream in this example is SIP/2712736XXXX and the transitive closure of the network in that direction.