[asterisk-users] One Way Audio Problem

[Tzafrir Cohen]

On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote:
> Hello Tzafrir,
> 
> On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
> >
> > This means Zaptel gets silence from Asterisk.
> >
> > What codecs are used? What do you see on 'sip show channels'?
> 
> I am using the following codecs:
> 
> # asterisk -rx 'sip show settings' | grep Codecs
>   Codecs:                 0xe (gsm|ulaw|alaw)
> 
> Below is the CLI output:
> 
>     -- Executing [91234567 at family:1] Dial("SIP/102-081d11d0",
> "Zap/4/1234567") in new stack
>     -- Called 4/1234567
> 
> *CLI> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format
>  Hold     Last Message
> 192.168.101.102  102         3c27a6824ba  00101/00002  0x4 (ulaw)
>  No       Rx: INVITE
> 1 active SIP channel
> 
> *CLI> core show channels
> Channel              Location             State   Application(Data)
> Zap/4-1              91234567 at inbound_tr Dialing AppDial((Outgoing Line))
> SIP/102-081d11d0     91234567 at family:1   Ring    Dial(Zap/4/1234567)
> 2 active channels
> 1 active call

So the call is not established yet, right?

This is not a temporary state?

> 
> > Can you call from the FXO to Asterisk? (e.g.: to echo test)
> 
> There is no problem with an inbound calls. I just tried to call the
> echo test extension number from my mobile phone via FXO/POTS and it
> works fine. I can hear my own voice.

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir