I have setup a SIP server and connected two polycom soundstation duo devices to it. They both register line with the SIP server fine but when I try to place a call from one device to the other, asterisk's CLI returns following NOTICE:
Call from '1000' (18.104.22.168:5060) to extension '1001' rejected because extension not found in context 'polycom'.
Related snipped of extension.conf look like:
[polycom] exten => 1000,1,Dial(SIP/1000) exten => 1000,2,VoiceMail,u1000 exten => 1000,102,VoiceMail,b1000 exten => 1001,1,Dial(SIP/1001) exten => 1001,2,VoiceMail,u1001 exten => 1001,104,VoiceMail,b1001
and sip.conf look like:
 type=friend host=dynamic dtmfmode=rfc2833 context=polycom  type=friend host=dynamic dtmfmode=rfc2833 context=polycom
Can you spot any error in these configurations?