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