Nasir Javaid
2010-May-13 14:30 UTC
[asterisk-users] asterisk-users Digest, Vol 70, Issue 30
sorry, you r right i just checked it with registration so there were astdb entries for SIP registration. anyhow after clearing settings frm astdb i tried the same scenario you advised but no luck. I think i told that i am not using server as peer but want to use a user [abc] as peer so that when ever i use dial(SIP/${EXTEN}@abc) or dial(SIP/abc/${EXTEN}) the call will be out from server using [abc]'s account. i hope you understand what i mean. also i will like to know is there any way that i can include registration information in my dial string so that i have no need to write register => abc:mysecred at nasir.server.com:8060 regards, Nasir Javaid Look, you do again with registration. remove any registration information. Look this config, I think it can help you Server1: sip.conf [interboxserver2] type=friend host=192.168.250.112 context=callfromserver2 disallow=all allow=ulaw allow=alaw allow=g729 extensions.conf [calltoserver2] exten => _X.,1,Noop(Call to server2) exten => _X.,2,Dial(SIP/ interboxserver2/${EXTEN}) exten => _X.,3,Hangup [callfromserver2] exten => _X.,1,Noop(Call from server2) exten => _X.,2,Dial(SIP/${EXTEN}) exten => _X.,3,Hangup Server2: sip.conf [interboxserver1] type=friend host=192.168.250.111 context=callfromserver1 disallow=all allow=ulaw allow=alaw allow=g729 extensions.conf [calltoserver1] exten => _X.,1,Noop(Call to server1) exten => _X.,2,Dial(SIP/interboxserver1/${EXTEN}) exten => _X.,3,Hangup [callfromserver1] exten => _X.,1,Noop(Call from server1) exten => _X.,2,Dial(SIP/${EXTEN}) exten => _X.,3,Hangup Try so, I think it must work. And also, look and delete any another records in both servers in sip.conf about this servers settings. Vardan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100513/01e7a560/attachment.htm