Crippled voicemail prompt (and other prompts)

Hi,

I have a Linksys SPA 3102 connected to FreePBX 2.8 with Asterisk 1.6.0.19
This Linksys uses ALAW codec and can make calls without any voice quality issues.

However, when I call any service on the PBX (like voicemail) , the prompts are played with cripppled and long sound.

I see that it is playing file vm-password.alaw on the console, but the sound is terrible. What is the reason for that ?

 -- Executing [[email protected]:1] Set("SIP/995-00000002", "AMPUSER=995") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/995-00000002", "0?report") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/995-00000002", "1?Set(REALCALLERIDNUM=995)") in new stack

    -- Executing [[email protected]:4] Set("SIP/995-00000002", "AMPUSER=995") in new stack

    -- Executing [[email protected]:5] Set("SIP/995-00000002", "AMPUSERCIDNAME=995") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/995-00000002", "0?report") in new stack

    -- Executing [[email protected]:7] Set("SIP/995-00000002", "AMPUSERCID=995") in new stack
    -- Executing [[email protected]:8] Set("SIP/995-00000002", "CALLERID(all)="995" <995>") in new stack

    -- Executing [[email protected]:9] ExecIf("SIP/995-00000002", "0?Set(CHANNEL(language)=)") in new stack

    -- Executing [[email protected]:10] GotoIf("SIP/995-00000002", "0?continue") in new stack
    -- Executing [[email protected]:11] Set("SIP/995-00000002", "__TTL=64") in new stack

    -- Executing [[email protected]:12] GotoIf("SIP/995-00000002", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [[email protected]:19] NoOp("SIP/995-00000002", "Using CallerID "995" <995>") in new stack

    -- Executing [*[email protected]:4] Macro("SIP/995-00000002", "get-vmcontext,995") in new stack

    -- Executing [[email protected]:1] Set("SIP/995-00000002", "VMCONTEXT=default") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/995-00000002", "0?200:300") in new stack
    -- Goto (macro-get-vmcontext,s,300)

    -- Executing [[email protected]:300] NoOp("SIP/995-00000002", "") in new stack

    -- Executing [*[email protected]:5] Set("SIP/995-00000002", "VMBOXEXISTSSTATUS=SUCCESS") in new stack
    -- Executing [*[email protected]:6] GotoIf("SIP/995-00000002", "1?mbexist") in new stack

    -- Goto (access,*00,106)
    -- Executing [*[email protected]:106] VoiceMailMain("SIP/995-00000002", "[email protected]") in new stack

    -- <SIP/995-00000002> Playing 'vm-password.alaw' (language 'en')

I’d appreciate if anyone could give a comment.

What have you tried so far?

Have your tried a SIP phone or a soft phone to see if the sound quality is better.

I tried Eyebeam softphone and Linksys SPA 3102.
The problem is not actually a voice quality problem. If I call a remote destination or another extension, I can hear audio on both weays with no quality issues.
The problem just occurs on prompts played by PBX itself. Description of the problem is voice stretch , words are getting long.

problem solved after re-installing asterisk 1.6.0.19
weird…