phone can't call

p2ii

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#1
I'm trying to get my first phone up and running to start testing. It is an Aastra 9133 and it appears registered but when I attempt to call anywhere inside I receive "the number you have dialed is not in service, please check the number and try again" In the log it shows that 104 tried to dial"s" instead of 555(voicemail)A little stumped. Thanks.


Forgot to add, my Xlite softphone is working fine.
 

p2ii

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#2
--Executing always shows the call coming from "external" from the Aastra 9133i instead of "internal" which is what shows when I make calls from xlite x100. The xlite calls go through fine. Also, I can call x104 just fine from the xlite x100.

this is the output in terminal when trying to call x100 from x104

-- Executing [100@from-sip-external:1] NoOp("SIP/104-08af59c0", "Received incoming SIP connection from unknown peer to 100" ) in new stack
-- Executing [100@from-sip-external:2] Set("SIP/104-08af59c0", "DID=100" ) in new stack
-- Executing [100@from-sip-external:3] Goto("SIP/104-08af59c0", "s|1" ) in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/104-08af59c0", "0?from-trunk|100|1" ) in new stack
-- Executing [s@from-sip-external:2] Set("SIP/104-08af59c0", "TIMEOUT(absolute)=15" ) in new stack
-- Channel will hangup at 2008-09-13 05:23:12 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/104-08af59c0", "" ) in new stack
-- Executing [s@from-sip-external:4] Wait("SIP/104-08af59c0", "2" ) in new stack
-- Executing [s@from-sip-external:5] Playback("SIP/104-08af59c0", "ss-noservice" ) in new stack
-- <SIP/104-08af59c0> Playing 'ss-noservice' (language 'en' )
== Spawn extension (from-sip-external, s, 5) exited non-zero on 'SIP/104-08af59c0'
-- Executing [h@from-sip-external:1] NoOp("SIP/104-08af59c0", "Hangup" ) in new stack
-- Executing [h@from-sip-external:2] Set("SIP/104-08af59c0", "DID=s" ) in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/104-08af59c0", "s|1" ) in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/104-08af59c0", "0?from-trunk|s|1" ) in new stack
-- Executing [s@from-sip-external:2] Set("SIP/104-08af59c0", "TIMEOUT(absolute)=15" ) in new stack
-- Channel will hangup at 2008-09-13 05:23:17 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/104-08af59c0", "" ) in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/104-08af59c0'
 

telecomtechnician

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#3
Hi there

1) Check very carefully the configuration of the aastra phone, specially the sip proxy or outbound proxy. Compare it to the xlite.

2) Did you create both extensions correctly through the elastix interface?

3) If you did, then do the following, register the xlite with the information of the aastra phone extension. If the xlite register with no problem and you can access voicemail, then your problem is the configuration of the aastra phone.

Waiting for your comments
David Medina
 

p2ii

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#4
Thanks for the reply. I ended up getting the phone up. It was a config problem. There are multiple places to put the same info in the web gui and I was putting it in the wrong place.
 

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