Re: [asterisk-users] Linksys not Ringing
I do not have any answer int he dialplan. what I mean is that when I call any other SIP phone is does the answer in the CLI. Even if I put and answer() in the dialplan still no ringing Jason Luki wrote: shouldn't there be an answer in there somewhere?... like... No... you can (and probably should) Dial() an extension before answering the incoming call. Do a sip debug and see if the Sipura is getting the INVITE message (and responding with an ACK), and if it sends back a RINGING message. Something strange is going here, and my bet is on some kind of NAT screw-up. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys not Ringing
I have 2 linksys SIP phones SPA-942 I have a dialplan of exten = 144,1,Wait(1) exten = 144,2,Dial(Sip/phil,20) exten = 144,3,Voicemail([EMAIL PROTECTED],u) The CLI looks like this when I dial 144 -- Executing Wait(IAX2/JASONSERVER-9, 1) in new stack -- Executing Dial(IAX2/JASONSERVER-9, Sip/phil|20) in new stack -- Called phil -- Nobody picked up in 2 ms -- Executing VoiceMail(IAX2/JASONSERVER-9, [EMAIL PROTECTED]|u) in new stack -- Playing 'vm-theperson' (language 'en') It is registered and will make calls but I never get the -- SIP/phil is ringing This happening on my 2 linksys phones only Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys not Ringing
jason,
shouldn't there be an answer in there somewhere?... like...
[inbound-sip]
exten = 300,1,Wait(1)
exten = 300,n,Answer()
exten = 300,n,NoOp(${EXTEN})
exten = 300,n,NoOp(${CALLERID})
exten = 300,n,Dial(SIP/300,15)
exten = 300,n,VoiceMailMain
exten = 300,n,Hangup()
daveC
Jason Walker wrote:
I
have 2 linksys SIP phones SPA-942
I have a dialplan of
exten = 144,1,Wait(1)
exten = 144,2,Dial(Sip/phil,20)
exten = 144,3,Voicemail([EMAIL PROTECTED],u)
The CLI looks like this when I dial 144
-- Executing Wait("IAX2/JASONSERVER-9", "1") in new stack
-- Executing Dial("IAX2/JASONSERVER-9", "Sip/phil|20") in new stack
-- Called phil
-- Nobody picked up in 2 ms
-- Executing VoiceMail("IAX2/JASONSERVER-9", "[EMAIL PROTECTED]|u") in
new stack
-- Playing 'vm-theperson' (language 'en')
It is registered and will make calls but I never get the
-- SIP/phil is ringing
This happening on my 2 linksys phones only
Jason
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Re: [asterisk-users] Linksys not Ringing
shouldn't there be an answer in there somewhere?... like... No... you can (and probably should) Dial() an extension before answering the incoming call. Do a sip debug and see if the Sipura is getting the INVITE message (and responding with an ACK), and if it sends back a RINGING message. Something strange is going here, and my bet is on some kind of NAT screw-up. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
