Hi I set the debug to 15, and changed to peer, I got this:
============================================================================ ============================================ <--- SIP read from UDP:94.77.210.xxx:5060 ---> SIP/2.0 500 account has been moved to a remote system Via: SIP/2.0/UDP 78.93.40.xxx:5060;branch=z9hG4bK637acf57 From: "8111488569" <sip:8111488...@78.93.40.xxx>;tag=as26c20707 To: <sip:0505103...@94.77.210.xxx>;tag=FF6EE3B3 Call-ID: 430bffa53f86bccf7566a3f9325a0...@78.93.40.xxx:5060 CSeq: 103 INVITE Server: CommuniGatePro/5.3.12d Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Got SIP response 500 "account has been moved to a remote system" back from 94.77.210.xxx:5060 Transmitting (no NAT) to 94.77.210.xxx:5060: ACK sip:0505103...@94.77.210.xxx SIP/2.0 Via: SIP/2.0/UDP 78.93.40.xxx:5060;branch=z9hG4bK637acf57 Max-Forwards: 70 From: "8111488569" <sip:8111488...@78.93.40.xxx>;tag=as26c20707 To: <sip:0505103...@94.77.210.xxx>;tag=FF6EE3B3 Contact: <sip:8111488...@78.93.40.xxx:5060> Call-ID: 430bffa53f86bccf7566a3f9325a0...@78.93.40.xxx:5060 CSeq: 103 ACK User-Agent: FPBX-2.8.1(1.8.7.0) Content-Length: 0 ============================================================================ ============================================ -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Monday, March 05, 2012 4:51 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] configure my voip provider Le 05/03/2012 07:37, Baha @ SH a écrit : > > Hello all, > > I am trying to configure my voip provider on asterisk box, I cannot > get the right configuration after trying many possibilities, always > getting circuits busy ,message, please let me know the meaning of the > debug message. > > And by the way, when I use a normal sip phone, I can dial normally, no > problems. > > First my config: > What is the [...] part? > host=xx.xx.xx.xx > > username=8111488569 > > secret=abcd > > fromuser=8111488569 > > type=user > type=peer as you call them. > allow=ulaw&alaw&gsm&g726 > > canreinvite=no > > dtmfmode=inband > > qualify=1000 > > insecure=very > > I also have registeration string: > > 8111488569:a...@xx.xx.xx.xx/8111488569 > > And my status on asterisk is registered. > > My debug: > > Mar 5 04:42:04 > > > > VERBOSE > > > > [20042] pbx.c: > > > > -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1001-00000004", > "SIP/8111488569/0505103250,300,") in new stack > > Mar 5 04:42:04 > > > > VERBOSE > > > > [20042] netsock2.c: > > > > == Using SIP RTP TOS bits 184 > > Mar 5 04:42:04 > > > > VERBOSE > > > > [20042] netsock2.c: > > > > == Using SIP RTP CoS mark 5 > > Mar 5 04:42:04 > > > > ERROR > > > > [20042] netsock2.c: > > > > getaddrinfo("8111488569", "(null)", ...): Name or service not known > > Mar 5 04:42:04 > > > > WARNING > > > > [20042] chan_sip.c: > > > > No such host: 8111488569 > You call your account number, not your provider IP Regards -- Daniel -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users