configs/voicemail.conf.sample - Make mailcmd sample text more explicit

Made the wording a bit more explicit. Didn't really change the meaning.
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Merged revisions 408876 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 408877 from http://svn.asterisk.org/svn/asterisk/branches/11
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git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408879 f38db490-d61c-443f-a65b-d21fe96a405b
res_pjsip_send_to_voicemail: transferring to voicemail for digium phones

Added the ability for transferring directly to voicemail on digium phones.
Added a new module that checks for the presence of a custom header and/or
diversion header within a sip REFER.  If either is found and they specify
a sending to voicemail action then variables are added to the channel
allowing the user access to them in the dialplan.  Dialplan can then be
written that branches based upon these values allowing, for instace, for
a single number to be used for dialing and/or accessing voicemail directly.

Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip
channels through (checked to make sure it has the correct channel type before
proceeding).

Review: https://reviewboard.asterisk.org/r/3245/
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Merged revisions 408880 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408881 f38db490-d61c-443f-a65b-d21fe96a405b
res_pjsip_exten_state: Presence for digium phones

Added presence support for digium phones.

Review: https://reviewboard.asterisk.org/r/3239/
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Merged revisions 408882 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: http://svn.asterisk.org/svn/asterisk/trunk@408883 f38db490-d61c-443f-a65b-d21fe96a405b