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Summary:ASTERISK-10116: Asterisk 1.4.9 and after crash when I check messages from peers behind NAT
Reporter:Ognjen Seslija (sekil)Labels:
Date Opened:2007-08-18 13:40:39Date Closed:2011-06-07 14:00:24
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Applications/app_voicemail
Versions:Frequency of
Occurrence
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Issues:
Environment:Attachments:
Description:I have several phones behind NAT that access my server. When I check messages by calling certain vm extension, asterisk crashes. Everything is working ok, when I try message check from phones in local network.


****** ADDITIONAL INFORMATION ******

from extensions.conf:

exten => 15000,1,VoicemailMain(${CALLERID(num)}|s)

debug prior to crash:

DEBUG[32665] pbx.c: Launching 'VoiceMailMain'
DEBUG[32665] app_voicemail.c: Before ast_answer
DEBUG[32665] devicestate.c: Notification of state change to be queued on device/channel SIP/502-08211d10
DEBUG[32665] chan_sip.c: SIP answering channel: SIP/502-08211d10
DEBUG[32612] devicestate.c: No provider found, checking channel drivers for SIP - 502
DEBUG [32612] chan_sip.c: Checking device state for peer 502
DEBUG[32612] devicestate.c: Changing state for SIP/502 - state 1 (Not in use)
DEBUG[32665] chan_sip.c: Setting framing from config on incoming call
DEBUG[32667] app_queue.c: Device 'SIP/502' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
DEBUG[32665] chan_sip.c: ** Our capability: 0x10c (ulaw|alaw|g729) Video flag: True
DEBUG[32612] devicestate.c: No provider found, checking channel drivers for SIP - 502
DEBUG[32665] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
DEBUG[32612] chan_sip.c: Checking device state for peer 502
DEBUG[32665] chan_sip.c: -- Done with adding codecs to SDP
DEBUG[32665] channel.c: Internal timing is enabled (option_internal_timing=1048576 chan->timingfd=21)
DEBUG[32665] chan_sip.c: Done building SDP. Settling with this capability: 0x10c (ulaw|alaw|g729)
DEBUG[32665] sched.c: Attempted to delete nonexistent schedule entry 0!
DEBUG[32665] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id  ASTERISK-217
DEBUG[32665] app_voicemail.c: After vm_authenticate
Comments:By: Joshua C. Colp (jcolp) 2007-08-19 20:03:51

Please try the latest version, 1.4.10.1 and if this issue still exists please follow the instructions in backtrace.txt in the doc directory.

By: Joshua C. Colp (jcolp) 2007-08-30 16:10:49

I'm suspending this due to lack of feedback, and I highly suspect this is fixed in the latest 1.4. If not feel free to grab the needed information and reopen.