[asterisk-bugs] [JIRA] Commented: (ASTERISK-18955) Voicemail message recording from IAX source sped up and jittery
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Tue Sep 25 09:35:27 CDT 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-18955?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197568#comment-197568 ]
Matt Jordan commented on ASTERISK-18955:
----------------------------------------
{quote}
Why is this bug still unfixed in a stable branch after 9 months when you have all this information and traces?
{quote}
As of September 25th, 2012, at 9:33 CST, there are 750 open issues in the ASTERISK project.
The answer to that question should be obvious: this issue will be worked by someone as time and developer resources become available.
If that response is not sufficient, patches are always welcome. If a patch is provided on an issue, it typically is resolved quicker than issues without patches (although that is a generalization - if a patch does not conform to the coding guidelines, is not well tested, or generally does not appear to fix the issue, then the presence of the patch doesn't really change the response time much).
If that response is not sufficient, please feel free to contact an Asterisk developer on the asterisk-biz list. There may be a developer who is willing to resolve your issue for you.
Thanks!
> 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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