Crippled voicemail prompt (and other prompts)

General Help
1

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 ?

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

    -- Executing [s@macro-user-callerid:4] Set("SIP/995-00000002", "AMPUSER=995") in new stack

    -- Executing [s@macro-user-callerid:5] Set("SIP/995-00000002", "AMPUSERCIDNAME=995") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/995-00000002", "0?report") in new stack

    -- Executing [s@macro-user-callerid:7] Set("SIP/995-00000002", "AMPUSERCID=995") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/995-00000002", "CALLERID(all)="995" <995>") in new stack

    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/995-00000002", "0?Set(CHANNEL(language)=)") in new stack

    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/995-00000002", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/995-00000002", "__TTL=64") in new stack

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

    -- Executing [*00@access:4] Macro("SIP/995-00000002", "get-vmcontext,995") in new stack

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

    -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/995-00000002", "") in new stack

    -- Executing [*00@access:5] Set("SIP/995-00000002", "VMBOXEXISTSSTATUS=SUCCESS") in new stack
    -- Executing [*00@access:6] GotoIf("SIP/995-00000002", "1?mbexist") in new stack

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

    -- <SIP/995-00000002> Playing 'vm-password.alaw' (language 'en')
3

I’d appreciate if anyone could give a comment.

4

What have you tried so far?

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

5

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.

6

problem solved after re-installing asterisk 1.6.0.19
weird…