[Asterisk-Users] SIP peer doesn't report busy properly

[Florian Overkamp]

Hi Remco, 

> -----Original Message-----
> I'm using wengo for my outgoing calls (SIP). However, 
> whenever a number is 
> busy, asterisk plays a message that the number you dialed is not 
> available instead of a busy signal.
> 
> How can I get the 'normal' PSTN tones (like number not in use 
> tone or busy 
> tone etc)
> 
> Or is this not possible with SIP?

Yes, it is possible to signal busy. However, the signalling must be correct
through the entire path, so that also includes a requirement for your
provider (Wengo). You can start by checking what signal they return to you
by checking the DIALSTATUS variable after a Dial.

Best regards,
Florian