[Asterisk-Users] asteisk, sip & NAT

[Hervé Thibaud]

Le dim 22/06/2003 à 12:18, Dan a écrit :

> exten => _8XXXXX,1,SetCallerID(${FWDUSERID})
> exten => _8XXXXX,2,SetCIDName(${FWDUSERNAME})
> exten => _8XXXXX,3,Dial(SIP/${EXTEN:1}@fwd.pulver.com)
> exten => _8XXXXX,4,Playback(invalid)
> exten => _8XXXXX,5,Hangup

It is better now, i try to call an other sip user (an other station but with sjphone directly registered to fwd)
the call ring but when i accept i have no sound
i try the other way it is not better, i have a sound with many blank
i have tested my sound card so that i registered radio on internet and registers are ok (on both)
it suppose that demonstrate my sound cards are full-duplex.

My connexion is an ISDN 64k/b and i suppose it's enough


Andy, your update is 
http://www.automated.it/guidetoasterisk.htm isn't it ?

i have an error when i start asterisk in :
 chan_modem.so (Generic Voice Modem Driver)
-- Parsing "/etc/asterisk/modem.conf': Found
-- Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulates Modem
Driver)
Warning(32771): File chan_oss.c Line 228 (sound_thread): Read error on
sound device; Ressource temporarily unavilable
---------------
I suppose this pb has matter with PSTN phone (tests was OK for me )

thanks