Choppy audio playback in voicemail

Asterisk Asterisk Support
1

Hi,
I’m having problems with the audio playback when calling the voicemail extention (6050) from any other extention. The audio is choppy and unrecognisable.

I can see the following in the console, so it appears I am reaching the voicemail extention ok.

-- Executing [6050@numberplan-custom-1:1] VoiceMailMain("SIP/6000-08215850", "") in new stack -- <SIP/6000-08215850> Playing 'vm-login' (language 'en')

I first thought perhaps the box is too slow, but it is a 800mhz with 512Mb ram, 100mb/s connection to the router, with only the default asteriskNow services running. I have read that asterisk should run ok on hardware much slower - so I don’t think this is the issue.

Then I thought it could be a nat issue (as I am behind a router with nat), but I am able to call between extentions both inside and outside nat (via asertisk) and the audio quality is fine.

I then though it could be a transcoding issue, perhaps when playing back the wav files. But when issuing “core show translation” almost everything is sub 10 miliseconds, only speex and ilbc are around 60ms (but I don’t think I am using them anyway).

I have been reading everything I can get my hands on, but havn’t been able to figure it out.

Any help would be greatly appreciated.

2

Thanks to Mathew, I have managed to fix this.

In the asteriskNow GUI, goto system configuration on the top right, then manage services and then stop services like there was no tomorrow!

I stopped:
ip6tables (don’t stop this if you need a firewall)
iptables (don’t stop this if you need a firewall)
netfs (don’t stop this if you need network shares)
nfs (don’t stop this if you need network shares)
ntpd (dont’ stop this if you need to sync time)
open-iscsi (dont’ stop this if you have scsi drives)
smb (don’t stop this if you need network shares)
zaptel (don’t stop this if you need zaptel hardware stuff)

And now the audio quality is fine.

3

or meetme or reliable iax channels.

4

Thanks for the info…
I’m new to asterisk (and having to learn quick), luckly i’m not using IAX or meetme - so hopefuly there shouldn’t be any sideeffects to stopping zaptel

5

We had problems with the Budgetone 101’s (hardphones) having horrible sounding voice. Resolved it by forcing the phone to only use uLaw and aLaw, but deselecting any other codecs as possibilities.

There were several threads in here about choppiness in the menus, but most of those people were running virtual machines. The choppiness was almost alway caused by timing issues.

forums.digium.com/viewtopic.php? … choppy+ivr

and

forums.digium.com/viewtopic.php? … choppy+ivr

These guys even have a kernel hack to set the timing that they said fixed the issue for them and was referred to in several forums

blogtech.oc9.com/index.php?optio … t&Itemid=1

For us, just moving from the VirtualBox to a dedicated machine fixed the problem

I see you already fixed yours with freeing some resources, but maybe this can help some wayfaring wanderer
Jop

6

Hi,
I’ve also had bad experiances with having asterisk in a virtual machine, but when trying to register sip devices - I kept getting rtp errors… I think it may of been to do with using a hacked version of vmware (to support a wifi card), that caused mac addressing issues (but thats only a theory).

You mention timming issues, what do you mean (i’m a noob)?