[asterisk-dev] Strange problem with new asterisk.

[Mathieu Rene]

You're getting a Permission Denied error, the dsp symlink is owned by
root, check permissions of /dev/dsp0 to make sure user 'asterisk' can
has rw permissions

Mathieu

On 10/20/06, Jonson Player <jonsonplayer at gmail.com> wrote:
> Hello,
> thank you for your reply.
>
> h-gw:/ # ls -la /dev/dsp
> lrwxrwxrwx  1 root root 4 Aug  6 15:35 /dev/dsp -> dsp0
>
> h-gw:/ #lsof -P -i -n
> ---
> ----
> asterisk  18464 asterisk    8u  IPv4 4196867       UDP *:4569
> asterisk  18464 asterisk    9u  IPv4 4196871       UDP *:2727
> asterisk  18464 asterisk   15u  IPv4 4196874       UDP *:5060
> asterisk  18464 asterisk   16u  IPv4 4196875       TCP *:2000 (LISTEN)
> asterisk  18464 asterisk   17u  IPv4 4196882       UDP *:4520
> ----
> ---
>
> On the other hand when i call 1000 (demo-test) - i cannot hear anything...
>
> Thank you for support.
>
>
> On 10/20/06, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
> > On Fri, Oct 20, 2006 at 05:27:35PM +0300, Jonson Player wrote:
> > > Hello i recently installed Asterisk 1.4.0-beta2. I hev some problems
> with
> > > local calls:
> > >
> > > ------------------<Cut Here>-----------------
> > > [Oct 20 17:25:09] WARNING[15655]: chan_oss.c:686 setformat: Unable to
> > > re-open DSP device /dev/dsp: Permission denied
> >
> > ls -l /dev/dsp
> >
> > Under what user does Asterisk run?
> >
> > >    -- Executing [1010 at default:1] Dial("OSS/dsp", "SIP/1010|40|t") in new
> > > stack
> > >    -- Called 1010
> > > [Oct 20 17:25:09] WARNING[15686]: channel.c:2965
> > > ast_channel_make_compatible: No path to translate from
> > > SIP/1010-081a9b60(256) to OSS/dsp(64)
> > >    -- SIP/1010-081a9b60 is ringing
> > >    -- SIP/1010-081a9b60 answered OSS/dsp
> > > [Oct 20 17:25:15] WARNING[15686]: channel.c:2965
> > > ast_channel_make_compatible: No path to translate from OSS/dsp(64) to
> > > SIP/1010-081a9b60(256)
> > > [Oct 20 17:25:15] WARNING[15686]: app_dial.c:1580 dial_exec_full: Had to
> > > drop call because I couldn't make OSS/dsp compatible with
> SIP/1010-081a9b60
> > >  == Spawn extension (default, 1010, 1) exited non-zero on 'OSS/dsp'
> > > << Hangup on console >>
> > > ------------------<And Here>-----------------
> > >
> > > Can someone help me?
> > > Thank you.
> >
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-dev mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> >
> > --
> > Tzafrir Cohen         sip:tzafrir at local.xorcom.com
> > icq#16849755          iax:tzafrir at local.xorcom.com
> > +972-50-7952406          jabber:tzafrir at jabber.org
> > tzafrir.cohen at xorcom.com     http://www.xorcom.com
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
>