Hi, I am trying to make external sip calls by using asterisk. Please provide information regarding sip trunk configuration in conf files. Setup is as below, * * *Case A:* Register two soft phones [X-lite] with 1000 and 10001 numbers to asterisk PBX [running in 192.168.1.11] and able to make calls in between. Sip.conf ===== [general] context=default bindport=5060 bindaddr=192.168.1.11 srvlookup=yes [1000] type=friend nat=yes host=dynamic canreinvite=no context=default allow=ulaw [1001] type=friend nat=yes host=dynamic canreinvite=no context=default allow=ulaw extensions.conf =========== [default] exten => 1000,1,Dial(SIP/1000) exten => 1001,1,Dial(SIP/1001) *Case B:* I have register other phone with ondo sip server running on other PC [192.168.1.12] with number as 6001. Now, Want to make calls between this two [asterisk PBX [1000/1001] on 192.168.1.11 and ondo [6001] on 192.168.1.12]. Please suggest how to configure sip trunk in conf files. Thanks in advance. Regards, Garge. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100616/901f6294/attachment.htm