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asterisk at solutionen... Guest
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Posted: Mon May 16, 2016 3:04 pm Post subject: [asterisk-users] Asterisk 11 on Centos: Voicemail crashes wh |
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Hi folks,
I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log.
The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check this). uname -a returns:
Linux asterisk.sjssolutions.local 3.10.0-327.13.1.el7.x86_64 #1 SMP Thu Mar 31 16:04:38 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux
On the CLI, I get this: Quote: |
== Using SIP RTP CoS mark 5
== Extension Changed 5103[hints] new state InUse for Notify User 5104
== Extension Changed 5103[hints] new state InUse for Notify User 5103
-- Executing [5106@internal:1] NoOp("SIP/5103-00000000", "-- Calling SJS extension 5106 from SIP/5103-00000000, transferring context") in new stack
-- Executing [5106@internal:2] Goto("SIP/5103-00000000", "sjs_extensions,5106,1") in new stack
-- Goto (sjs_extensions,5106,1)
-- Executing [5106@sjs_extensions:1] Dial("SIP/5103-00000000", "IAX2/remoteAsterisk/5106,10") in new stack
-- Called IAX2/remoteAsterisk/5106
-- Call accepted by <ip_address_of_remoteAsterisk> (format ulaw)
-- Format for call is (ulaw)
-- IAX2/remoteAsterisk-17114 is ringing
-- IAX2/remoteAsterisk-17114 is ringing
-- Nobody picked up in 10000 ms
-- Hungup 'IAX2/remoteAsterisk-17114'
-- Executing [5106@sjs_extensions:2] VoiceMail("SIP/5103-00000000", "5103,u") in new stack
[May 16 20:37:58] WARNING[14514][C-00000000]: res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
[May 16 20:37:58] WARNING[14514][C-00000000]: res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
[May 16 20:37:58] WARNING[14514][C-00000000]: res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
> 0x7f61e008b750 -- Probation passed - setting RTP source address to 10.0.0.190:5004
-- <SIP/5103-00000000> Playing 'vm-theperson.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/5.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/1.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/0.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/3.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'vm-isunavail.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'vm-intro.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'beep.ulaw' (language 'en_GB')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5103/tmp/nzuoKd format: wav, 0x7f621800bba8
asterisk*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups |
(note: Yes, it's deliberate that it's going to a different extension VM... the call goes via another asterisk server to a remote phone; then comes back if unanswered to record the VM.)
The system starts to create the file, and sometimes even records some bytes, before dying: Quote: |
[root@asterisk tmp]# ls -l
total 4
-rw-r--r-- 1 root root 0 May 16 20:38 nzuoKd
-rw-r--r-- 1 root root 44 May 16 20:38 nzuoKd.wav |
Note: I've since changed the safe_asterisk script to start up Asterisk as asterisk:asterisk, it seems to still work; apart from VM which crashes the same way.
I tried setting the file format to ulaw, this had the same problem (except the temp file ended with .ulaw). I saw a similar problem had been solved in version 1.6.1, except that didn't seem to show the "x=0, open writing:" message.
System has plenty of available disk space (40G or 179G depending on which bit of the filesystem you look at).
I've never seen this on any of the Asterisk servers I've run (many, since v1.4), but I mostly run it on Ubuntu variants, this is my first Centos...
Addendum: I modified safe_asterisk & got the following when it quit:
/usr/sbin/safe_asterisk: line 163: 18115 Illegal instruction (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Any ideas gratefully received. I'm going to try installing a compiled-from-source version of 13.7 at the weekend, can't do it before then as it's our production office system... Everything apart from VM seems to work (although if anyone can shed any light on the frequent "res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short" warnings I'm seeing, that'd also be appreciated.
Oh - one more thing, I had to disable 2 codecs (lpc10 and ilbc) because they used an instruction that doesn't exist on the server (it's an oldish HP mini-server). I'm guessing from the above message that VM might be afflicted by the same issue. Presumably compiling from source will solve this? (I've compiled 13.7, no errors reported, but I've not tried running it yet)
Cheers!
Ade. |
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brian at wildsong.biz Guest
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Posted: Mon May 16, 2016 3:08 pm Post subject: [asterisk-users] Asterisk 11 on Centos: Voicemail crashes wh |
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Did you build from source on one machine and install on another? I ran into something like that, have to turn off optimizations in the build environment if you do that and the machine architecture is different.
On Mon, May 16, 2016 at 1:03 PM, asterisk <asterisk@solutionengineers.com (asterisk@solutionengineers.com)> wrote:
Quote: |
Hi folks,
I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log.
The system is running on Centos 6 (or maybe 6.5, I'm not sure how to check this). uname -a returns:
Linux asterisk.sjssolutions.local 3.10.0-327.13.1.el7.x86_64 #1 SMP Thu Mar 31 16:04:38 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux
On the CLI, I get this: Quote: |
== Using SIP RTP CoS mark 5
== Extension Changed 5103[hints] new state InUse for Notify User 5104
== Extension Changed 5103[hints] new state InUse for Notify User 5103
-- Executing [5106@internal:1] NoOp("SIP/5103-00000000", "-- Calling SJS extension 5106 from SIP/5103-00000000, transferring context") in new stack
-- Executing [5106@internal:2] Goto("SIP/5103-00000000", "sjs_extensions,5106,1") in new stack
-- Goto (sjs_extensions,5106,1)
-- Executing [5106@sjs_extensions:1] Dial("SIP/5103-00000000", "IAX2/remoteAsterisk/5106,10") in new stack
-- Called IAX2/remoteAsterisk/5106
-- Call accepted by <ip_address_of_remoteAsterisk> (format ulaw)
-- Format for call is (ulaw)
-- IAX2/remoteAsterisk-17114 is ringing
-- IAX2/remoteAsterisk-17114 is ringing
-- Nobody picked up in 10000 ms
-- Hungup 'IAX2/remoteAsterisk-17114'
-- Executing [5106@sjs_extensions:2] VoiceMail("SIP/5103-00000000", "5103,u") in new stack
[May 16 20:37:58] WARNING[14514][C-00000000]: res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
[May 16 20:37:58] WARNING[14514][C-00000000]: res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
[May 16 20:37:58] WARNING[14514][C-00000000]: res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short
> 0x7f61e008b750 -- Probation passed - setting RTP source address to 10.0.0.190:5004
-- <SIP/5103-00000000> Playing 'vm-theperson.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/5.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/1.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/0.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'digits/3.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'vm-isunavail.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'vm-intro.ulaw' (language 'en_GB')
-- <SIP/5103-00000000> Playing 'beep.ulaw' (language 'en_GB')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/5103/tmp/nzuoKd format: wav, 0x7f621800bba8
asterisk*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups |
(note: Yes, it's deliberate that it's going to a different extension VM... the call goes via another asterisk server to a remote phone; then comes back if unanswered to record the VM.)
The system starts to create the file, and sometimes even records some bytes, before dying: Quote: |
[root@asterisk tmp]# ls -l
total 4
-rw-r--r-- 1 root root 0 May 16 20:38 nzuoKd
-rw-r--r-- 1 root root 44 May 16 20:38 nzuoKd.wav |
Note: I've since changed the safe_asterisk script to start up Asterisk as asterisk:asterisk, it seems to still work; apart from VM which crashes the same way.
I tried setting the file format to ulaw, this had the same problem (except the temp file ended with .ulaw). I saw a similar problem had been solved in version 1.6.1, except that didn't seem to show the "x=0, open writing:" message.
System has plenty of available disk space (40G or 179G depending on which bit of the filesystem you look at).
I've never seen this on any of the Asterisk servers I've run (many, since v1.4), but I mostly run it on Ubuntu variants, this is my first Centos...
Addendum: I modified safe_asterisk & got the following when it quit:
/usr/sbin/safe_asterisk: line 163: 18115 Illegal instruction (core dumped) nice -n $PRIORITY "${ASTSBINDIR}/asterisk" -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY}
Any ideas gratefully received. I'm going to try installing a compiled-from-source version of 13.7 at the weekend, can't do it before then as it's our production office system... Everything apart from VM seems to work (although if anyone can shed any light on the frequent "res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short" warnings I'm seeing, that'd also be appreciated.
Oh - one more thing, I had to disable 2 codecs (lpc10 and ilbc) because they used an instruction that doesn't exist on the server (it's an oldish HP mini-server). I'm guessing from the above message that VM might be afflicted by the same issue. Presumably compiling from source will solve this? (I've compiled 13.7, no errors reported, but I've not tried running it yet)
Cheers!
Ade.
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Brian Wilson, GISP
Wildsong 707-827-0001 |
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asterisk at solutionen... Guest
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Posted: Tue May 17, 2016 4:58 am Post subject: [asterisk-users] Asterisk 11 on Centos: Voicemail crashes wh |
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Hi Brian,
The version I'm running right now was installed from RPMs, so I guess that counts as built on one machine & installed on another...
I've already compiled v13.7, next step is to try it out & see if that fixes the issue.
Thanks for your reply!
Cheers,
Ade.
On 16/05/2016 21:08, Brian Wilson wrote:
Quote: | Did you build from source on one machine and install on another? I ran into something like that, have to turn off optimizations in the build environment if you do that and the machine architecture is different.
On Mon, May 16, 2016 at 1:03 PM, asterisk <asterisk@solutionengineers.com (asterisk@solutionengineers.com)> wrote:
Quote: |
Hi folks,
I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7 LTS), I've just configured the voicemail function, and it's mostly working fine... except when I try to leave a voicemail! This crashes asterisk with no entries in the messages log.
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