Did you try it the magic number of times, three?

On Sun, Oct 12, 2008 at 9:57 PM, GNUbie <[EMAIL PROTECTED]> wrote:
> Hello Tzafrir,
>
> On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED]: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              [EMAIL PROTECTED] Dialing AppDial((Outgoing Line))
> SIP/102-081d11d0     [EMAIL PROTECTED]: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
>
> _______________________________________________
> -- 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
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

_______________________________________________
-- 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

Reply via email to