I've been using the wiki as a guide here...
http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net
I've also tried various permutations, combinations and contortions but nada.
below is my extensions.conf
***** begin extensions.conf ****
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
[globals]
CONSOLE = Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO = guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK = Zap/G2 ; Trunk interface
TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:p...@provider
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[macro-voicemail]
exten => s,1,Dial(${ARG1},20)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten => s-BUSY,n,Hangup()
exten => _s-.,1,Goto(s-NOANSWER,1)
[incoming_calls]
exten => ekiga_meyamma_in,1,Macro(voicemail,SIP/101)
exten => ekiga_meyamma_in,n,Hangup()
[internal_calls]
exten => 101,1,Macro(voicemail,SIP/101)
exten => 101,n,Hangup()
exten => 102,1,Dial(SIP/102)
exten => 102,n,Hangup()
exten => 8,1,VoiceMailMain(s${CALLERIDNUM})
exten => 8,n,Hangup()
exten => 600,1,Answer()
exten => 600,n,Playback(demo-echotest)
exten => 600,n,Echo()
exten => 600,n,Playback(demo-echodone)
exten => 600,n,Hangup()
[outgoing_calls]
exten => _9.,1,Dial(SIP/ekiga_meyamma_out/${EXTEN:1},20,r))
exten => _9.,n,Hangup()
[home]
include => internal_calls
include => outgoing_calls
***** end extensions.conf ****
below is my sip.conf
**** begin sip.conf ***
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
allow=h261
videosupport=yes
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions
register => meyamma:meya...@ekiga.net/ekiga_meyamma_in
externhost=fspublic.selfip.com
externrefresh=10 ;
localnet=192.168.1.0/255.255.255.0
[authentication]
;setup
[101]
type=friend
username=101
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=home
;port=5061
;setup
[102]
type=friend
username=102
secret=welcome
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=home
;port=5061
;ekiga.net
[ekiga_meyamma_out]
type=peer
username=meyamma
secret=meyamma
host=ekiga.net
canreinvite=no
qualify=300
insecure=port,invite
;ekiga.net
[ekiga_meyamma_in]
type=user
username=meyamma
secret=meyamma
host=ekiga.net
canreinvite=no
qualify=300
context=incoming_calls
insecure=port,invite
**** end sip.conf ****
These are the latest incantations of my sip and extensions conf files. Can
anyone assist in helping me to get incoming calls from ekiga.net set up. I
have the outgoing working as that was the simple part. That one is handled by
[ekiga_meyamma_out] and [outgoing_calls].
Thanks a mil
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