[asterisk-users] One Way Audio Problem

[GNUbie]

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

> 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.

Thank you.

Regards,

GNUbie