Hi, this is the second time that i post this, may be a wasnt clear the first time. Im having problems with an incoming peer after i upgraded asterisk from 1.0 to 1.2.4, in 1.0 i used to configure the incoming peers like this: register => @prepago-in [prepago-in] type=friend host=192.168.10.102 ; this is the cisco's ip context = from-external dtmfmode=rfc2833 insecure=very ; required for incoming FWD calls in cisco as5400 the dial-peer is configured like this: dial-peer voice 2662 voip tone ringback alert-no-PI description OUTPUT_TO_ASTERISK translation-profile outgoing remove_# destination-pattern 22662[0,1,8]T voice-class codec 5 session protocol sipv2 session target ipv4:192.168.10.103 <--- this is the asterisk's ip dtmf-relay rtp-nte Using asterisk v1.0 i can receive calls perfectly, after i upgraded to asterisk v1.2.4 i receive the following error Feb 28 16:49:34 WARNING[11142]: chan_sip.c:3207 sip_register: Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line 154 Of course, i cant receive incoming call anymore, reading the error i undestand that im missing the username in the register => line in sip.conf , as you can see, there is not username parameter in the cisco's dial-peer configuration. Is the username a required parameter in 1.2.4, if so, why did you do this change? any help help will be greatly appreciated thanks ---- Miguel