Hello there . I have been facing a problem in my Asterisk. I have TDM400 card with 4 port. 2 FXO + 2 FXS. When I call to SIP extension 2000 from FXS1 and try to transfer that call to SIP 2003 extension, it transfer…everything is OK. But the problem is if any of SIP phone you disconnect from the software X-Lite / Zoiper, it is not disconnect…you can see in the Flash operator panel those SIP are still connected. Line is not free. So what should I do now ? Please help me.
If you add something “Dial” command in [from-internal] context and Hangup(), I found a problem there. Try to avoid Dial() + Hangup() application in there.
Voice Problem :
If anybody has voice problem in SIP phone please read this documents carefully. It will work 100%. My one worked properly.
Mohammed Hasan (Rupam)