[asterisk-dev] How do you test when you develop dialplan applications?

[Paolo Ornati]

On Mon, 1 Oct 2007 10:19:05 +0200
lumen <lumen at bulma.net> wrote:

> My environment to test are two softphones installed on my own machine, and
> the asterisk server in that machine too.

Probably the best thing is to have SIP hardphones.

Mine problems with softphones were that they "conflicted" with each
other.

I solved that with Ekiga under Linux/X11 in this way:

1) create some dummy users: voip1 voidp2 voip3 ...

2) configure ekiga for the different users to listen on different ports:
	gconf-editor --> apps --> Ekiga --> ... --> sip --> listen_port

example:
	user	port
	voip1	5061
	voip2	5062
	voip3	5063

3) close every instance of Ekiga and start Asterisk

4) start the soft-phones with a script like this (I assume sudo is
well configured):
---------------------------------------
#!/bin/bash
USERS="voip1 voip2 voip3"

for U in $USERS; do
        echo "start $U"
        xauth extract - $DISPLAY | sudo -H -u "$U" xauth merge -
        sudo -b -H -u "$U" env DISPLAY="$DISPLAY" ekiga
        sleep 1
done
---------------------------------------

PS: to avoid troubles with the different instances of Ekiga complaining
about the sound card (everyone tries to access it) I suggest to load
the "snd_dummy" driver and use that...

-- 
	Paolo Ornati
	Linux 2.6.23-rc8-ga64314e6 on x86_64