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On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts...<BR>
<BR>
Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre.<BR>
Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung)<BR>
<BR>
When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers.<BR>
When I only activate the first account, then the first account registers well to the public Asterisk-server on the internet.<BR>
When I only activate the second, then the second account registers well to the local Asterisk-server (NSLU unslung).<BR>
<BR>
Is it normal that I can not use both accounts at the same time ?! One local and one to a public server ??<BR>
<BR>
<BR>
When only the first account is enabled on the Grandstream IP-telephone, then the local Asterisk-server CLI shows this (when SIP debugging) :<BR>
<BR>
<FONT SIZE="2">---</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:21] Really destroying SIP dialog '0d8ad63d6c1940a11ff1ee23728468ba@192.168.1.77' Method: OPTIONS</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:31] Reliably Transmitting (no NAT) to 192.168.1.100:5064:</FONT><BR>
<FONT SIZE="2">OPTIONS sip:nslu@192.168.1.100:5064;transport=udp SIP/2.0</FONT><BR>
<FONT SIZE="2">Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport</FONT><BR>
<FONT SIZE="2">From: "asterisk" <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>>;tag=as1cad824f</FONT><BR>
<FONT SIZE="2">To: <<A HREF="sip:nslu@192.168.1.100:5064">sip:nslu@192.168.1.100:5064</A>;transport=udp></FONT><BR>
<FONT SIZE="2">Contact: <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>></FONT><BR>
<FONT SIZE="2">Call-ID: <A HREF="mailto:64b39841529a74ad44a0b44403287993@192.168.1.77">64b39841529a74ad44a0b44403287993@192.168.1.77</A></FONT><BR>
<FONT SIZE="2">CSeq: 102 OPTIONS</FONT><BR>
<FONT SIZE="2">User-Agent: Asterisk-jocan</FONT><BR>
<FONT SIZE="2">Max-Forwards: 70</FONT><BR>
<FONT SIZE="2">Date: Sat, 10 Oct 2009 16:09:31 GMT</FONT><BR>
<FONT SIZE="2">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT><BR>
<FONT SIZE="2">Supported: replaces</FONT><BR>
<FONT SIZE="2">Content-Length: 0</FONT><BR>
<BR>
<BR>
<FONT SIZE="2">---</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:32] Retransmitting #1 (no NAT) to 192.168.1.100:5064:</FONT><BR>
<FONT SIZE="2">OPTIONS sip:nslu@192.168.1.100:5064;transport=udp SIP/2.0</FONT><BR>
<FONT SIZE="2">Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport</FONT><BR>
<FONT SIZE="2">From: "asterisk" <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>>;tag=as1cad824f</FONT><BR>
<FONT SIZE="2">To: <<A HREF="sip:nslu@192.168.1.100:5064">sip:nslu@192.168.1.100:5064</A>;transport=udp></FONT><BR>
<FONT SIZE="2">Contact: <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>></FONT><BR>
<FONT SIZE="2">Call-ID: <A HREF="mailto:64b39841529a74ad44a0b44403287993@192.168.1.77">64b39841529a74ad44a0b44403287993@192.168.1.77</A></FONT><BR>
<FONT SIZE="2">CSeq: 102 OPTIONS</FONT><BR>
<FONT SIZE="2">User-Agent: Asterisk-jocan</FONT><BR>
<FONT SIZE="2">Max-Forwards: 70</FONT><BR>
<FONT SIZE="2">Date: Sat, 10 Oct 2009 16:09:31 GMT</FONT><BR>
<FONT SIZE="2">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT><BR>
<FONT SIZE="2">Supported: replaces</FONT><BR>
<FONT SIZE="2">Content-Length: 0</FONT><BR>
<BR>
<BR>
<FONT SIZE="2">---</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:33] Retransmitting #2 (no NAT) to 192.168.1.100:5064:</FONT><BR>
<FONT SIZE="2">OPTIONS sip:nslu@192.168.1.100:5064;transport=udp SIP/2.0</FONT><BR>
<FONT SIZE="2">Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport</FONT><BR>
<FONT SIZE="2">From: "asterisk" <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>>;tag=as1cad824f</FONT><BR>
<FONT SIZE="2">To: <<A HREF="sip:nslu@192.168.1.100:5064">sip:nslu@192.168.1.100:5064</A>;transport=udp></FONT><BR>
<FONT SIZE="2">Contact: <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>></FONT><BR>
<FONT SIZE="2">Call-ID: <A HREF="mailto:64b39841529a74ad44a0b44403287993@192.168.1.77">64b39841529a74ad44a0b44403287993@192.168.1.77</A></FONT><BR>
<FONT SIZE="2">CSeq: 102 OPTIONS</FONT><BR>
<FONT SIZE="2">User-Agent: Asterisk-jocan</FONT><BR>
<FONT SIZE="2">Max-Forwards: 70</FONT><BR>
<FONT SIZE="2">Date: Sat, 10 Oct 2009 16:09:31 GMT</FONT><BR>
<FONT SIZE="2">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT><BR>
<FONT SIZE="2">Supported: replaces</FONT><BR>
<FONT SIZE="2">Content-Length: 0</FONT><BR>
<BR>
<BR>
<FONT SIZE="2">---</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:34] Retransmitting #3 (no NAT) to 192.168.1.100:5064:</FONT><BR>
<FONT SIZE="2">OPTIONS sip:nslu@192.168.1.100:5064;transport=udp SIP/2.0</FONT><BR>
<FONT SIZE="2">Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport</FONT><BR>
<FONT SIZE="2">From: "asterisk" <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>>;tag=as1cad824f</FONT><BR>
<FONT SIZE="2">To: <<A HREF="sip:nslu@192.168.1.100:5064">sip:nslu@192.168.1.100:5064</A>;transport=udp></FONT><BR>
<FONT SIZE="2">Contact: <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>></FONT><BR>
<FONT SIZE="2">Call-ID: <A HREF="mailto:64b39841529a74ad44a0b44403287993@192.168.1.77">64b39841529a74ad44a0b44403287993@192.168.1.77</A></FONT><BR>
<FONT SIZE="2">CSeq: 102 OPTIONS</FONT><BR>
<FONT SIZE="2">User-Agent: Asterisk-jocan</FONT><BR>
<FONT SIZE="2">Max-Forwards: 70</FONT><BR>
<FONT SIZE="2">Date: Sat, 10 Oct 2009 16:09:31 GMT</FONT><BR>
<FONT SIZE="2">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT><BR>
<FONT SIZE="2">Supported: replaces</FONT><BR>
<FONT SIZE="2">Content-Length: 0</FONT><BR>
<BR>
<BR>
<FONT SIZE="2">---</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:35] Retransmitting #4 (no NAT) to 192.168.1.100:5064:</FONT><BR>
<FONT SIZE="2">OPTIONS sip:nslu@192.168.1.100:5064;transport=udp SIP/2.0</FONT><BR>
<FONT SIZE="2">Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport</FONT><BR>
<FONT SIZE="2">From: "asterisk" <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>>;tag=as1cad824f</FONT><BR>
<FONT SIZE="2">To: <<A HREF="sip:nslu@192.168.1.100:5064">sip:nslu@192.168.1.100:5064</A>;transport=udp></FONT><BR>
<FONT SIZE="2">Contact: <<A HREF="sip:asterisk@192.168.1.77">sip:asterisk@192.168.1.77</A>></FONT><BR>
<FONT SIZE="2">Call-ID: <A HREF="mailto:64b39841529a74ad44a0b44403287993@192.168.1.77">64b39841529a74ad44a0b44403287993@192.168.1.77</A></FONT><BR>
<FONT SIZE="2">CSeq: 102 OPTIONS</FONT><BR>
<FONT SIZE="2">User-Agent: Asterisk-jocan</FONT><BR>
<FONT SIZE="2">Max-Forwards: 70</FONT><BR>
<FONT SIZE="2">Date: Sat, 10 Oct 2009 16:09:31 GMT</FONT><BR>
<FONT SIZE="2">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</FONT><BR>
<FONT SIZE="2">Supported: replaces</FONT><BR>
<FONT SIZE="2">Content-Length: 0</FONT><BR>
<BR>
<BR>
<FONT SIZE="2">---</FONT><BR>
<FONT SIZE="2">[Oct 10 18:09:35] Really destroying SIP dialog '64b39841529a74ad44a0b44403287993@192.168.1.77' Method: OPTIONS</FONT><BR>
<BR>
Why is there an 'option' send to the local Asterisk-server when the local account on the Grandstream is disabled ?!<BR>
<BR>
Thanks for showing me some insight in all this !<BR>
<BR>
Jonas.
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