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 ------------------------------------------------------------------------- Using Tomcat but need to do more? Need to support web services, security? Get stuff done quickly with pre-integrated technology to make your job easier Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 _______________________________________________ Sipp-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/sipp-users
