[Ekiga-list] Calling sip:500 ekiga net from Asterisk?
- From: Gilles <codecomplete free fr>
- To: ekiga-list gnome org
- Subject: [Ekiga-list] Calling sip:500 ekiga net from Asterisk?
- Date: Wed, 15 Dec 2010 07:51:35 -0800 (PST)
Hello
I'm an Asterisk newbie, and would like to call the free numbers at Ekiga to
perform echo tests, etc.
http://wiki.ekiga.org/index.php/Fun_Numbers
I managed to connect an SIP client to Asterisk, and Asterisk to a VoIP
provider, with the SIP client + Asterisk behind a NAT router. I can dial out
to my cellphone through my VOSP, and receive calls from my cellphone with
two-way audio just fine:
http://img560.imageshack.us/img560/3749/asterisknat.png
However, when I type "sip:500 ekiga net" into XLite, it says "Call
established" but I don't see anything in the Asterisk console (sip set
debug) and get no sound from the Ekiga server:
;=========== sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
srvlookup = yes
externip=<my public IP>
localnet=192.168.0.0/24
;all RTP packets go through Asterisk
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
register => myaccount:mypass myvosp com
[myvosp_outgoing]
type=peer
host=myvosp.com
username=myaccount
secret=mypass
fromuser=myaccount
fromdomain=myvosp.com
nat=yes
canreinvite=no
[myvosp_incoming]
type=peer
host=myvosp.com
context=from_myvosp
nat=yes
canreinvite=no
[6011]
type=friend
context=my-phones
secret=6011
host=dynamic
qualify=yes
;NOT NEEDED? nat=yes
nat=no
;=========== extensions.conf
[general]
static=yes
writeprotect=yes
clearglobalvars=no
autofallthrough=yes
[from_myvosp]
exten => s,1,Dial(SIP/6011)
exten => s,n,Hangup()
[my-phones]
exten => 6011,1,Dial(SIP/6011)
exten => 6011,n,Hangup()
exten => 6012,1,Dial(SIP/6012)
exten => 6012,n,Hangup()
exten => _0.,1,Dial(SIP/myvosp_outgoing/${EXTEN})
exten => _0.,n,Hangup()
;===========
Does someone know what I should put in sip.conf and extensions.conf to get
this working?
Thank you.
--
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