[Asterisk-Users] How to properly handle incoming SIP and IAX calls, so user can call back and how to properly make outgoing sip/iax calls through Asterisk ?

[Robert Rozman]

Hi,

I'm aware that incoming and outgoing calls are going fine when isdn channels 
are involved - caller id properly identifies calling party, so user can call 
back....

But how to properly handle this for iax, sip calls....

I have few questions :
- BTW, what to type for instance in remote firefly to make standalone calls 
to Asterisk default context or particular extension ?

- If I receive incoming sip or iax call and is then saved as for instance in 
Firefly. Now Firefly would like to call back that caller, but call goes not 
through Asterisk... Why ? How to do this properly?

- Outogoing calls: how to properly send outgoind iax or sip calls through 
asterisk, so each calling extension gets proper caller id, so can be called 
back.... ?

Any experience or existing solution to this problem? Any advice ?

Regards,

Rob. 

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