> Darryl Moore <[email protected]> schrieb:
>
> > I'd start by turning on sip debugging in asterisk
> > >sip set debug ip [your_phone_ip]
>
> Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.
> 16.34.133' Method: OPTIONS
> Reliably Transmitting (no NAT) to 192.168.200.11:5060:
> OPTIONS sip:[email protected]:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport
> Max-Forwards: 70
> From: "asterisk" <sip:[email protected]>;tag=as1215345d
> To: <sip:[email protected]:5060>
> Contact: <sip:[email protected]>
> Call-ID: [email protected]
> 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.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
