[asterisk-bugs] [JIRA] Commented: (ASTERISK-18955) Voicemail message recording from IAX source sped up and jittery

[seb7 (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-18955?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197545#comment-197545 ] 

seb7 commented on ASTERISK-18955:
---------------------------------

I have the same problem on the lastest CentOS Asterisk 1.8 on i686 on a fully up-to-date CentOS 5:
core show version
Asterisk 1.8.15.1 built by root @ 92-139-19-10.digium.internal on a i686 running Linux on 2012-08-31 18:57:50 UTC

If I leave a voicemail on the remote system over an IAX2 trunk, the message is saved with the audio sped up and jittery. If I turn off IAX2 trunking on the server that is sending the call to the remote system, the voicemail message is fine. If I turn off trunking only on the destination server it has no effect: the message is still speed up and jittery to the point of being *indecipherable*.

Why is this bug still unfixed in a stable branch after 9 months when you have all this information and traces?

> Voicemail message recording from IAX source sped up and jittery
> ---------------------------------------------------------------
>
>                 Key: ASTERISK-18955
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18955
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_voicemail, Channels/chan_iax2
>    Affects Versions: 1.8.7.1
>         Environment: Asterisk 1.8.7.1 i686 on CentOS 5
>            Reporter: Steven Premeau
>            Assignee: Leif Madsen
>         Attachments: console.log, iax.log, iax.okay.pcap, iax.pcap, menuselect.makedeps, menuselect.makeopts
>
>
> When asterisk records voice mail on a call received over a IAX connection (from a third party provider) the audio is sped up and jittery.
> Calls received by the trunk and forward to a Polycom SIP phone are fine.
> This appears to be the same issue reported in http://forums.asterisk.org/viewtopic.php?f=1&t=74149&p=145170 and possibly similar to ASTERISK-18330
> Given my connection to the net and NAT, switching to a SIP trunk is not the preferred solution.

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