Re: [Ekiga-list] Calling sip:500 ekiga net from Asterisk?

[Thierry Simonnet <t simonnet esiee fr>]

The best way to connect your private asterisk to ekiga.net is to use a 
trunk.
Then you will call ekiga.net numbers using (for example) a prefix:
sip:x500 myvosp com

Don't hesitate to send me a mail, I will give you the conf I use.

Best

Th. Simonnet

Le 15/12/2010 16:51, Gilles a écrit :
> 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.