On Tue, Sep 3, 2013 at 4:25 AM, S, Kantharuban IN MAA SL < kantharuba...@siemens.com> wrote:
> Hi List,**** > > The below error caused the Asterisk to crash, if anyone > have idea on this please reply,(Asterisk version :1.8.9)**** > > ** ** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel > SIP/18202-0002d011 setting write format to ilbc from ulaw native formats > 0x4 (ulaw)**** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec > translation path from 0x4 (ulaw) to 0x400 (ilbc)**** > > [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type > ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) > **** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel > SIP/18202-0002d011 setting write format to ilbc from ulaw native formats > 0x4 (ulaw)**** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec > translation path from 0x4 (ulaw) to 0x400 (ilbc)**** > > [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type > ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) > **** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel > SIP/18202-0002d011 setting write format to ilbc from ulaw native formats > 0x4 (ulaw)**** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec > translation path from 0x4 (ulaw) to 0x400 (ilbc)**** > > [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type > ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) > **** > > [Sep 2 15:59:53] WARNING[24418] channel.c: No path to translate from > SIP/18252-0002d010 to SIP/18203-0002d01e**** > > [Sep 2 15:59:53] WARNING[24418] channel.c: Can't make SIP/18252-0002d010 > and SIP/18203-0002d01e compatible**** > > [Sep 2 15:59:53] WARNING[24418] features.c: Bridge failed on channels > SIP/18252-0002d010 and SIP/18203-0002d01e**** > > [Sep 2 16:00:00] WARNING[4970] chan_dahdi.c: Ignoring any changes to > 'userbase' (on reload) at line 23.**** > > [Sep 2 16:00:00] WARNING[4970] chan_dahdi.c: Ignoring any changes to > 'vmsecret' (on reload) at line 31.**** > > [Sep 2 16:00:00] WARNING[4970] chan_dahdi.c: Ignoring any changes to > 'hassip' (on reload) at line 35.**** > > [Sep 2 16:00:00] WARNING[4970] chan_dahdi.c: Ignoring any changes to > 'hasiax' (on reload) at line 39.**** > > [Sep 2 16:00:00] WARNING[4970] chan_dahdi.c: Ignoring any changes to > 'hasmanager' (on reload) at line 47.**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: No valid transports available, > falling back to 'udp'.**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' > for a peer/user that differs from the global setting can make**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! the name of that peer/user > discoverable by an attacker. Replies for non-existent peers/users**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! will be sent to a > different port than replies for an existing peer/user. If at all possible, > **** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! use the global 'nat' > setting and do not set 'nat' per peer/user.**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! (config > category='analog-fxs-gateway' global force_rport='Yes' peer/user > force_rport='No')**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' > for a peer/user that differs from the global setting can make**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! the name of that peer/user > discoverable by an attacker. Replies for non-existent peers/users**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! will be sent to a > different port than replies for an existing peer/user. If at all possible, > **** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! use the global 'nat' > setting and do not set 'nat' per peer/user.**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! (config > category='S0-gateway' global force_rport='Yes' peer/user force_rport='No') > **** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! PLEASE NOTE: Setting 'nat' > for a peer/user that differs from the global setting can make**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! the name of that peer/user > discoverable by an attacker. Replies for non-existent peers/users**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! will be sent to a > different port than replies for an existing peer/user. If at all possible, > **** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! use the global 'nat' > setting and do not set 'nat' per peer/user.**** > > [Sep 2 16:00:00] WARNING[4970] chan_sip.c: !!! (config > category='S2M-gateway' global force_rport='Yes' peer/user force_rport='No') > **** > > [Sep 2 16:00:00] WARNING[4970] pbx.c: Extension '_0.', priority 42 in > 'outgoing', label 'UP' already in use at priority 33**** > > [Sep 2 16:00:00] WARNING[4970] pbx.c: Extension '_0.', priority 45 in > 'outgoing', label 'DOWN' already in use at priority 36**** > > [Sep 2 16:00:01] WARNING[5180] acl.c: Cannot connect**** > > ** ** > > > It looks like you performed a 'sip reload' during an active call and Asterisk crashed. There have been numerous bug fixes to the 1.8 and later branches to address this kind of issue - since you're running a version of Asterisk 1.8 that is 20 months old, there is a good likelihood that any issue you are facing has already been fixed. Upgrading to a more recent version of 1.8 may be your best course of action. In any case, a log file only showing WARNING messages is often not sufficient for debug a problem, much less a crash. There are explicit instructions on the Asterisk wiki [1] on how to obtain the correct information when Asterisk crashes. If this happens again, please obtain a backtrace using the instructions and file an issue on the Asterisk issue tracker [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira Thanks Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
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