Cant hear playbacks / voicemail only calls

Asterisk Asterisk Support
1

Hi All,

I have a weird issue poped up out of the sudden on my VPS:
asterisk 11.5.x (combined with freepbx)
centos 6.4 64x
VPS running in openvz, main server OS is Centos 5.8

I cant hear when asterisk is playing, however calls between extensions and out are working.
Additionally, i have a copy of the VPS container in earlier stage of configuration, i loaded this VPS container and same issue appearing there (though it had older freepbx and configurations).

I tried to change codecs: ulaw,alaw,gsm
tested also with Firewall / Fail2Ban off - diffrent voip devices & locations.
Incoming cals from regular phone directed to voicemail also can’t hear anything.

As an example, calling *43 for echo test :

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [*43@from-internal:1] Answer("SIP/1000-0000007d", "") in new stack
       > 0x2b613007f4e0 -- Probation passed - setting RTP source address to 176.226.31.22:4010
    -- Executing [*43@from-internal:2] Wait("SIP/1000-0000007d", "1") in new stack
       > 0x2b613007f4e0 -- Probation passed - setting RTP source address to 176.226.31.22:4010
    -- Executing [*43@from-internal:3] Playback("SIP/1000-0000007d", "demo-echotest") in new stack
    -- <SIP/1000-0000007d> Playing 'demo-echotest.ulaw' (language 'en')
  == Spawn extension (from-internal, *43, 3) exited non-zero on 'SIP/1000-0000007d'
    -- Executing [h@from-internal:1] Hangup("SIP/1000-0000007d", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1000-0000007d

Please let me know what info shall i attach to help me resolve this.

Thanks in advanced for your support.

2

Have you try using differents sound files like GSM or SLN. Too for debug purpose you can convert an mp3 file to SLN using sox and then try to play the file and check if you can hear it.

sox file.mp3 -t raw -r 8000 -s -2 -c 1 file.sln

3

I have tried to swich codecs order in freepbx and sip client side,
GSM,ulaw, alaw, g722 non worked.
It was working until a week ago.

I have tried
sox /var/lib/asterisk/sounds/en/demo-echotest.mp3 -t raw -r 8000 -s -2 -c 1 /var/lib/asterisk/sounds/en/demo-echotest.sln
But seems i don’t have mp3 version, ulaw and alaw only.

4

Take an mp3 file from your LOCAL PC, upload the file to your PBX and then convert the file using sox and try to playback that file.

5

I tried the sox, but seems i don;t have the support for mp3
so i downloaded from downloads.asterisk.org/pub/telephony/sounds/ sln16 pack

i changed in freepbx the order, to slin alaw ulaw gsm
on asterisk i see its playing ulaw version still
when i have the order:
gsm . slin > alaw… . then gsm file is playing

6

sln16 is 16kHz. It has to be rate adapted before it can be used with the common, 8kHz codecs.

7


sox file.mp3 -t raw -r 8000 -s -2 -c 1 file.sln

8

I managed to convert mp3 to sln, however as posted before in freepbx when i define codec order as
sln > gsm > ulaw then demo-echotest plays gsm version and not sln (though now i have demo-echotest.sln)

The SLN test took the fucos out of the original issue, that i cant hear the playback sounds, when a week ago it was working.

I also mentioned that i have additional VPS which is an older working copy and there i faced same issue.
which makes this a weird case…

  • not a firewall issue
  • not configurations issue (by testing the old frozed vps, i have same issue)
  • not files issue…
    then what left?

Additionally, on the copy VPS i ran the asterisk install again (make install) and no diffrences.

Meanwhile i will try to make asterisk play the SLN file and check if i hear anything

9

I think this may have strayed off the real question, so I’ll ask you what timing source are you using?

Regarding the codecs, are you setting the codecs in sip.conf? If so, the typical phone will not recognize sln, so, if they can will choose gsm. The sound file used is the one with the lowest cost to convert it to the format actually used by the phone.

10

After reading about timing i did a test and it has an interesting results:

timing test Attempting to test a timer with 50 ticks per second. Using the 'pthread' timing module for this test. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. poll() timed out! This is bad. It has been 1000 milliseconds, and we got 0 timer ticks

This seems to be bad :smiley:
I’m not sure what timing i use, it should be in modules.conf? as i don’t see anything related to timing there.

11

I think it is in asterisk.conf.

12

can’t find it asterisk.conf

[code][directories]
astetcdir => /etc/asterisk
astmoddir => /usr/lib64/asterisk/modules
astvarlibdir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
transmit_silence_during_record = yes
languageprefix=yes
execincludes=yes

[/code]

13

I think you need internal_timing=yes

The actual source might be set by menuconfig, when building.

14

Problem solved!!!
I had a big issue with my server hosted at OVH, since their support is so slow and they had holidays with long weekend, i migrated my server to other host.

I tested the voip VPS and sound came back.
So either OVH added kinda of filtering (though RTP wasnt blocked) or server had hardware issue? or was it part of american-french -snowden spy that affected? :smiley:

Thank you david55 for your support on this :wink: