[asterisk-users] Newbie: Two problems with Asterisk Config, Please Help

[Pete Kay]

Hi,
I can get the message recorded and played correctly with wengo, but not with
zoiper.  Is there any codec setting that I should fixed and how to fixed it?

On Fri, Mar 21, 2008 at 9:26 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

> Probably a codec issue.  SIP debug while making a call would be helpful.
>
> Thanks,
> Steve Totaro
>
> On Fri, Mar 21, 2008 at 4:06 AM, Pete Kay <petedao at gmail.com> wrote:
> > Hi,
> > I switched to Wengo and solved the one "beat"problem. However, I am
> still
> > not able to listen to the recorded .wav sound.  Can anyone please point
> me
> > to the right direction?  How to listen to the .wav sound?
> >
> > Thanks,
> > Pete
> >
> >
> >
> > On Fri, Mar 21, 2008 at 9:34 AM, Carlos Rojas <crt.rojas at gmail.com>
> wrote:
> > > Hello,
> > >
> > > Do your verify, the codecs, of both clients, in your sip.conf?
> > >
> > > What codec do you use?
> > >
> > > Best Regards
> > >
> > >
> > >
> > >
> > >
> > > On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay <petedao at gmail.com> wrote:
> > >
> > > >
> > > >
> > > >
> > > > Hi,
> > > > I am sorry my questinos are too fundamental.  I am new to Asterisk,
> and
> > hope to catch up as fast as I can.
> > > >
> > > > Problem 1:
> > > >
> > > > I have my SIP  client ( in one PC .102) and SIP server ( in another
> PC
> > .101) within the same land.  They can make SIP connection, but when the
> SIP
> > client makes call to play an audio file, I can only hear a "beat"
> sounds,
> > and then nothing else.  In the console, I can see:
> > > > *CLI>     -- Executing [111 at my-phones:1] Answer("SIP/2001-081dd6e0",
> "")
> > in new stack
> > > >     -- Executing [111 at my-phones:2] VoiceMail("SIP/2001-081dd6e0",
> > "2000") in new stack
> > > > Sent RTP packet to      58.251.75.228:9956 (type 00, seq 037718, ts
> > 000160, len 000160)
> > > >     -- <SIP/2001-081dd6e0> Playing 'vm-intro' (language 'en')
> > > > Sent RTP packet to      58.251.75.228:9956 (type 00, seq 037719, ts
> > 000320, len 000160)
> > > > Sent RTP packet to      58.251.75.228:9956 (type 00, seq 037720, ts
> > 000480, len 000160)
> > > > Sent RTP packet to      58.251.75.228:9956 (type 00, seq 037721, ts
> > 000640, len 000160)
> > > > Got  RTP packet from    192.168.1.102:8000 (type 00, seq 062222, ts
> > 1373137124, len 000160)
> > > > Sent RTP packet to      192.168.1.102:8000 (type 00, seq 037722, ts
> > 000800, len 000160)
> > > > Sent RTP packet to      192.168.1.102:8000 (type 00, seq 037723, ts
> > 000960, len 000160)
> > > >
> > > > Is it the prolem?  First it sends to the public address of the the
> > router, then it sends to the virtual IP.  Is this the problem that
> causing
> > my to hear just one "beat" sound and then no audio?
> > > >
> > > > Problem 2:
> > > >
> > > > The problem is isolated from Problem 1, cuz I run the SIP client on
> the
> > same machine as the server, so there should not be network problem.  I
> > recorded some voice mails and they are stored as .wav files ok.  When I
> > tried to hear back the message, It does not work.  Is there any
> > configuration that I have to go through to have Asterisk to play .wav
> file?
> > > >
> > > > Thank you very much in advance for all your kind help.
> > > >
> > > > Pete
> > > >
> > > >
> > > > _______________________________________________
> > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > _______________________________________________
> >  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> >  asterisk-users mailing list
> >  To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080322/7edd6a3b/attachment.htm