would be nice if you could put this howto in the wiki (http://sipp.sourceforge.net/wiki/)
Thanks!
Olivier.
On 11/10/06, Juan Antonio Alvarez <[EMAIL PROTECTED]> wrote:
Hi again.
I can give you a hand if you need some help getting your scenario
working on asterisk.
I got it working doing what I said on the first item. i.e, telling
Asterisk the fixed location for both UAC and UAS. What's not so
realistic about this test, is that the server is not processing
registers nor INVITEs authentication. But I guess it can get you going
to try some things.
Anyway. This is a mini-howto that I wrote originally in Spanish. I'm
translating it here (so please excuse my English!):
----------------------------
Asterisk's sip.conf
Write the next lines in your asterisk's sip.conf file. Of course,
replace IPs, ports, etc as you need...
[sippuac]
type=friend
username=sippuac
host=192.168.101.9
port=5060
context=test
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
[sippuas]
type=friend
username=sippuas
host=192.168.101.9
port=5061
context=test
dtmfmode=rfc2833
insecure=very
canreinvite=no
nat=yes
Small explanation:
Two UAs are created. One will be the client, that generates calls. The
other one, the UAS, will recieve them.
Both UA can run in the same PC. That's why in the example they have
same host address. Which application will receive it, is known by the
port number. The parameter insecure=very makes asterisk not to require
authorization from the UAC when attempting to make a call. (If you get
it working with auth, please complete this howto, I didn't have time
to work on that yet).
The parameter nat=yes is because otherwise, the UAS misses the call
ACK. Not quite sure why this happens, I understand that the UAS should
ACK the call to the endpoint, according to the standard, but as
asterisk changes the callids, that may get messy for sipp to
understand. So when you set nat=yes, asterisk stand in the middle all
the time. (PLEASE somebody explain this better if I'm too confused)
Finally, context=test, tells that the UA enters that context in
extensions.conf when calling.
On extensions.conf just add
[test]
exten=>s,1,Dial(SIP/sippuas,20)
Here, you just define the call context, so when "s" service is
required, the UAS is called.
Running SIPp
Now you should run two instances of sipp. One as a UAC and one as UAS.
./sipp -sn uas -p 5061 -mp 6001
./sipp -s s -sn uac -p 5060 -i 192.168.101.9 192.168.101.249
Of course, all parameters should fit with your network and config files.
Hope that helps. Please let me know!
Good luck,
Juan
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