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 <[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
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:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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