[asterisk-users] Peer is UNREACHABLE
Luca Bertoncello
lucabert at lucabert.de
Thu May 28 15:40:57 CDT 2015
Darryl Moore <darryl at moores.ca> schrieb:
> I'd start by turning on sip debugging in asterisk
> >sip set debug ip [your_phone_ip]
Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.200.11:5060:
OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 172.16.34.133>;tag=as1215345d
To: <sip:00493512222222 at 192.168.200.11:5060>
Contact: <sip:asterisk at 172.16.34.133>
Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6 at 172.16.34.133
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
Date: Thu, 28 May 2015 20:39:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
repeated in loop...
Help that?
192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 the IP of the Asterisk server.
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
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