[asterisk-bugs] [JIRA] (ASTERISK-25684) noise instead of audio on slin codec with translation

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25684?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228931#comment-228931 ] 

Asterisk Team commented on ASTERISK-25684:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> noise  instead of audio on slin codec with translation
> ------------------------------------------------------
>
>                 Key: ASTERISK-25684
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25684
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: 13.1.0, 13.4.0, 13.6.0
>         Environment: Ubuntu 14.4 i686 3.19.0-42 on VMware WS 10
> FreeBSD 10.2 i386 on VirtualBox 5.0.10
> Ubuntu 14.4 i686 3.19.0 on bare hw (intel core i5)
>            Reporter: Victor Sverdlin
>
> Then one peer is slin (L16/8000) or slin16 (L16/16000) and second peer is other codec (tested with ulaw, gsm and speex) slin peer listen only noise. Noise disappear if other peer muted. Slin peer can listen voice if signal is gained down by AGC set to 10 and less (anyway sound is distorted).
> Both peers use sip channel.
> Testes with several peers:
> - slin: MicroSIP 3.10.9, custom HW device
> - other: antiSIP 4.2.9 (Android), SFLphone 1.3.0, Zoiper 3.3.25608
> Asterisk compiled with gcc 4.8.5(FreeBSD) and 4.8.4(Ubuntu).
> On Ubuntu Asterisk compiled with DONT_OPTIMIZE option.
> RTP capture will be attached.



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