[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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