SPA942 randomly getting set on Calls Forwarded

I have 8 SPA942 and 2 SPA504G Linksys/Cisco IP phones on a freePBX phone system and they randomly get set to Calls Forwarded and the phones randomly get locked where it asks for a password to access call log etc.  I reset the phones to default reset all settings and the phones work, but still these issues come up, with or without a password being set on the phones, using firmware 6.1.5a on the SPA942 phones.
I have setup the phone system a second time from scratch and still this issue comes up.  I cannot find any reason why the phones would randomly get set to calls forwarded, nor can I find any evidence that someone is logging into the phones and setting or changing the password.  
Don't know where to go from here?
Sincerely,
Matt Wyckoff

Turn on syslog & debug and catch all the messages sent from the phone (or all phones). It will help you with the further analysis.

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    Patrick
    Location/Ubicacion: Arizona USA
    Time Zone/Hora Local: UTC/GMT -7
    If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
    Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
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    /* Style Definitions */
    table.MsoNormalTable
    {mso-style-name:"Table Normal";
    mso-tstyle-rowband-size:0;
    mso-tstyle-colband-size:0;
    mso-style-noshow:yes;
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    mso-para-margin:0cm;
    mso-para-margin-bottom:.0001pt;
    mso-pagination:widow-orphan;
    font-size:11.0pt;
    font-family:"Calibri","sans-serif";
    mso-ascii-font-family:Calibri;
    mso-ascii-theme-font:minor-latin;
    mso-fareast-font-family:"Times New Roman";
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    no supplementary-service sip refer
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    dtmf-relay rtp-nte
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    translation-profile outgoing SIPOUT
    destination-pattern 1T
    session protocol sipv2
    session target sip-server
    voice-class codec 1 
    voice-class sip dtmf-relay force rtp-nte
    voice-class sip profiles 101
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    no vad
    dial-peer voice 102 voip
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    destination-pattern 0[2-9].T
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    dtmf-relay rtp-nte
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    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
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    max-dn 12
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    calling-number initiator
    timeouts interdigit 5
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    /* Style Definitions */
    table.MsoNormalTable
    {mso-style-name:"Table Normal";
    mso-tstyle-rowband-size:0;
    mso-tstyle-colband-size:0;
    mso-style-noshow:yes;
    mso-style-priority:99;
    mso-style-qformat:yes;
    mso-style-parent:"";
    mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
    mso-para-margin:0cm;
    mso-para-margin-bottom:.0001pt;
    mso-pagination:widow-orphan;
    font-size:11.0pt;
    font-family:"Calibri","sans-serif";
    mso-ascii-font-family:Calibri;
    mso-ascii-theme-font:minor-latin;
    mso-fareast-font-family:"Times New Roman";
    mso-fareast-theme-font:minor-fareast;
    mso-hansi-font-family:Calibri;
    mso-hansi-theme-font:minor-latin;}
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  • Is there any other way to achieve per user call forward restriction other than to create multiple voice policies?

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    Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
    SWC Unified Communications
    This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

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    Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
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