[Asterisk-Users] No caller ID, straight to voicemail

Michael Delage onesmallstep at gmail.com
Fri Jul 22 12:36:12 MST 2005


Hi,

I am having a problem with inbound calls (from a SIP VIOP provider).  
When caller ID information is not available, the calls go straight to 
voicemail.  We are using a mix of either Sipura 841 phones or SPAs.

When the call is passed to the phone/SPA, Asterisk reports "Got SIP 
Response 406 "Not Acceptable" back from..."

I have searched a while now and can't seem to find any reference to the 
cause of this error.  Does anyone know what could be causing this?  Is 
it an Asterisk issue or a setting on the SPA?

Thanks,

Michael

The relevant section from the log file is here:

Jul 22 14:27:15 DEBUG[17654]: Call from user '201' is 1 out of 0
Jul 22 14:27:15 VERBOSE[17654]: -- Called 201
Jul 22 14:27:15 DEBUG[17654]: Driver for channel 
'SIP/64.26.157.252-09bab2b0' does not support indication 3, emulating it
Jul 22 14:27:15 DEBUG[17654]: Scheduling timer at 160 sample intervals
Jul 22 14:27:15 DEBUG[17654]: (Provisional) Stopping retransmission (but 
retaining packet) on '274770fe769a07f51085b97c2198fa4b at 69.196.249.124' 
Request 102: Found
Jul 22 14:27:15 DEBUG[17654]: Acked pending invite 102
Jul 22 14:27:15 DEBUG[17654]: Stopping retransmission on 
'274770fe769a07f51085b97c2198fa4b at 69.196.249.124' of Request 102: Found
Jul 22 14:27:15 VERBOSE[17654]: -- Got SIP response 406 "Not Acceptable" 
back from 192.168.1.11
Jul 22 14:27:15 DEBUG[17654]: update_user_counter(201) - decrement 
outUse counter
Jul 22 14:27:15 VERBOSE[17654]: == No one is available to answer at this 
time
Jul 22 14:27:15 DEBUG[17654]: Scheduling timer at 0 sample intervals
Jul 22 14:27:15 DEBUG[17654]: Exiting with DIALSTATUS=NOANSWER.




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