Hello I have been having issues with kamailio and asterisk realtime. I have used all the configurations posted, but it just has not worked for me. What I am trying to do is to use asterisk as PSTN & voicemail for kamailio. But I keep getting this 401 not authorized from asterisk like this: ======== --- (19 headers 19 lines) --- Sending to 99.89.26.17:5060 (NAT) Using INVITE request as basis request - LCklNT_HoIeTxCB_8cSIf9efRNvkcR -- adding dns manager for '99.89.26.17' Scheduling destruction of SIP dialog &#39;3d67147a652fe8641c536eb92383ad74(a)99.89.26.17&#39; in 32000 ms (Method: NOTIFY) Reliably Transmitting (NAT) to 99.89.26.17:5060: NOTIFY sip:1000@99.89.26.17 SIP/2.0 Via: SIP/2.0/UDP 99.89.26.18:5060;branch=z9hG4bK00545e8e;rport Max-Forwards: 70 From: "asterisk" <sip:1000@99.89.26.17>;tag=as4d28adb7 To: <sip:1000@99.89.26.17> Contact: <sip:1000@99.89.26.18:5060> Call-ID: 3d67147a652fe8641c536eb92383ad74(a)99.89.26.17 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:1@99.89.26.17 Voice-Message: 0/0 (0/0) --- Found peer '1000' for '1000' from 99.89.26.17:5060 <--- Reliably Transmitting (NAT) to 99.89.26.17:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 99.89.26.17;branch=z9hG4bK354f.e07b39b2.0;received=99.89.26.17;rport=5060 Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bKj4sndhbgkh863bu8fpr61tb;rport=5060 From: <sip:1000@99.89.26.17>;tag=5tnt79v6phhc689kd5vh To: <sip:+2348023098407@99.89.26.17;user=phone>;tag=as21d7a164 Call-ID: LCklNT_HoIeTxCB_8cSIf9efRNvkcR CSeq: 1710 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="99.89.26.17", nonce="16c11ac7" Content-Length: 0 ======== asterisk and kamailio are on different server, and I have put the IP of asterisk in trusted table in kamailio db. My kamailio.cfg is: