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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' (58.65.17.222:5060) 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:

[1000]
type=friend
host=dynamic
dtmfmode=rfc2833
context=polycom

[1001]
type=friend
host=dynamic
dtmfmode=rfc2833
context=polycom

Can you spot any error in these configurations?

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1 Answer 1

Your configuration files look fine. And pardon if this is stupid question, but - have you reloaded the dialplan? And what result does it have to add

exten=>s,1,Answer()
exten=>s,n,Playback(beep)
exten=>s,n,Hangup() 

this to to your polycom context?

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