[asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

> 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.
> 

The phone you gave your wife is really old. Are you sure it supports SIP 
OPTIONS? Can you make a call in or out to it? If you can, it is more 
likely that it just doesn't support that and you can't use a qualify 
statement.
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