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
'[email protected]' Method: OPTIONS
[Oct 10 18:09:31] Reliably Transmitting (no NAT) to 192.168.1.100:5064:
OPTIONS sip:[email protected]:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport
From: "asterisk" <sip:[email protected]>;tag=as1cad824f
To: <sip:[email protected]:5064;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport
From: "asterisk" <sip:[email protected]>;tag=as1cad824f
To: <sip:[email protected]:5064;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport
From: "asterisk" <sip:[email protected]>;tag=as1cad824f
To: <sip:[email protected]:5064;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport
From: "asterisk" <sip:[email protected]>;tag=as1cad824f
To: <sip:[email protected]:5064;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK769c6ff5;rport
From: "asterisk" <sip:[email protected]>;tag=as1cad824f
To: <sip:[email protected]:5064;transport=udp>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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
'[email protected]' 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.
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