jonas kellens
2009-Oct-10 16:13 UTC
[asterisk-users] Grandstream GXP 2010 : multiple accounts not working
On my Grandstream GXP 2010 I have the possibility for 6 channels and thus 6 different accounts... Line 1 I define an account that registers directly to an online Asterisk-server, somewhere in a datacentre. Line 2 I define an account that registers to the local Asterisk-server (NSLU2 unslung) When I activate both accounts, only the first account (to the Asterisk-server on the internet) registers. When I only activate the first account, then the first account registers well to the public Asterisk-server on the internet. When I only activate the second, then the second account registers well to the local Asterisk-server (NSLU unslung). Is it normal that I can not use both accounts at the same time ?! One local and one to a public server ?? When only the first account is enabled on the Grandstream IP-telephone, then the local Asterisk-server CLI shows this (when SIP debugging) : --- [Oct 10 18:09:21] Really destroying SIP dialog '0d8ad63d6c1940a11ff1ee23728468ba at 192.168.1.77' Method: OPTIONS [Oct 10 18:09:31] Reliably Transmitting (no NAT) to 192.168.1.100:5064: OPTIONS sip:nslu at 192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" <sip:asterisk at 192.168.1.77>;tag=as1cad824f To: <sip:nslu at 192.168.1.100:5064;transport=udp> Contact: <sip:asterisk at 192.168.1.77> Call-ID: 64b39841529a74ad44a0b44403287993 at 192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:32] Retransmitting #1 (no NAT) to 192.168.1.100:5064: OPTIONS sip:nslu at 192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" <sip:asterisk at 192.168.1.77>;tag=as1cad824f To: <sip:nslu at 192.168.1.100:5064;transport=udp> Contact: <sip:asterisk at 192.168.1.77> Call-ID: 64b39841529a74ad44a0b44403287993 at 192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:33] Retransmitting #2 (no NAT) to 192.168.1.100:5064: OPTIONS sip:nslu at 192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" <sip:asterisk at 192.168.1.77>;tag=as1cad824f To: <sip:nslu at 192.168.1.100:5064;transport=udp> Contact: <sip:asterisk at 192.168.1.77> Call-ID: 64b39841529a74ad44a0b44403287993 at 192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:34] Retransmitting #3 (no NAT) to 192.168.1.100:5064: OPTIONS sip:nslu at 192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" <sip:asterisk at 192.168.1.77>;tag=as1cad824f To: <sip:nslu at 192.168.1.100:5064;transport=udp> Contact: <sip:asterisk at 192.168.1.77> Call-ID: 64b39841529a74ad44a0b44403287993 at 192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:35] Retransmitting #4 (no NAT) to 192.168.1.100:5064: OPTIONS sip:nslu at 192.168.1.100:5064;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport From: "asterisk" <sip:asterisk at 192.168.1.77>;tag=as1cad824f To: <sip:nslu at 192.168.1.100:5064;transport=udp> Contact: <sip:asterisk at 192.168.1.77> Call-ID: 64b39841529a74ad44a0b44403287993 at 192.168.1.77 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Sat, 10 Oct 2009 16:09:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Oct 10 18:09:35] Really destroying SIP dialog '64b39841529a74ad44a0b44403287993 at 192.168.1.77' Method: OPTIONS Why is there an 'option' send to the local Asterisk-server when the local account on the Grandstream is disabled ?! Thanks for showing me some insight in all this ! Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/8abff9ab/attachment-0001.htm
Gordon Henderson
2009-Oct-10 16:58 UTC
[asterisk-users] Grandstream GXP 2010 : multiple accounts not working
On Sat, 10 Oct 2009, jonas kellens wrote:> On my Grandstream GXP 2010 I have the possibility for 6 channels and > thus 6 different accounts... > > Line 1 I define an account that registers directly to an online > Asterisk-server, somewhere in a datacentre. > Line 2 I define an account that registers to the local Asterisk-server > (NSLU2 unslung) > > When I activate both accounts, only the first account (to the > Asterisk-server on the internet) registers. > When I only activate the first account, then the first account registers > well to the public Asterisk-server on the internet. > When I only activate the second, then the second account registers well > to the local Asterisk-server (NSLU unslung). > > Is it normal that I can not use both accounts at the same time ?! One > local and one to a public server ??The times I've seen this fail have been when the phone is behind a router with a broken SIP ALG. Make sure the ALG is turned off, if possible, and look into using a STUN server to let them phone know how the NAT firewall is working if the remote isn't behind a proxy. And if you are already using STUN, make sure the phone isn't trying to use STUN for the internal connection - it's programmable on a per-account basis in the GXP phones. Gordon