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asterisk

2016-May-16 20:03 UTC

[asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

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:

       == 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 at internal:1] NoOp("SIP/5103-00000000",
"--
    Calling SJS extension 5106 from SIP/5103-00000000, transferring
    context") in new stack
         -- Executing [5106 at internal:2] Goto("SIP/5103-00000000",
    "sjs_extensions,5106,1") in new stack
         -- Goto (sjs_extensions,5106,1)
         -- Executing [5106 at 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 at 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:

    [root at 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

2016-May-16 20:08 UTC

head link

[asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

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 at
solutionengineers.com>
wrote:
> 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:
>
>   == 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 at internal:1] NoOp("SIP/5103-00000000",
"-- Calling
> SJS extension 5106 from SIP/5103-00000000, transferring context") in
new
> stack
>     -- Executing [5106 at internal:2] Goto("SIP/5103-00000000",
> "sjs_extensions,5106,1") in new stack
>     -- Goto (sjs_extensions,5106,1)
>     -- Executing [5106 at 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 at 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:
>
> [root at 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.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Brian Wilson, GISP
Wildsong 707-827-0001
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asterisk

2016-May-17 09:57 UTC

head link

[asterisk-users] Asterisk 11 on Centos: Voicemail crashes when recording message

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:> 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 at solutionengineers.com 
> <mailto:asterisk at solutionengineers.com>> wrote:
>
>     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.
>
><snip>
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