On Tue, Sep 3, 2013 at 4:25 AM, S, Kantharuban IN MAA SL <
kantharuban.s at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130903/2751d859/attachment.htm>