[asterisk-dev] [Code Review] Adjust to allow for Digium phones' send to voicemail feature

[Mark Michelson]

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/certified/branches/1.8.11/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/1925/#comment11559>

    This change is actually revision 366597 of the 1.8 branch. However, since that change had not been merged into the 1.8.11 certified branch, I had to add it to this diff.


- Mark


On May 17, 2012, 9:54 a.m., Mark Michelson wrote:
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> (Updated May 17, 2012, 9:54 a.m.)
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> 
> Review request for Asterisk Developers, Jason Parker and Matt Jordan.
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> Summary
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> Digium phones have a couple of ways to send calls to voicemail.
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> On an incoming call, a call may be diverted to either the user's own, or to another user's, voicemail using the "send to vm" softkey.
> On an bridged call, a user may select a contact and press the "transfer vm" key in order to blind transfer the call to the contact's voicemail.
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> Contrary to the way this is likely done in other Asterisk installations, sending a call to a contact's voicemail does not send the call to a different extension than would be used for dialing the contact. Instead, something must be available in the dialplan to distinguish an incoming call that is intended for a phone vs. an incoming call that is intended for voicemail.
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> The mechanism by which this is accomplished is via a Diversion header's reason parameter. When a call is being sent to voicemail, the reason parameter will be set to "send_to_vm".
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> For the first case, on a redirected incoming call, the Diversion header is in the 302 response sent from the Digium phone.
> The second case is a bit unorthodox, but from our readings, not harmful or non-compliant. In the second case, the Diversion header is in the REFER request sent from the Digium phone when it performs its blind transfer.
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> The changes to Asterisk are minimal here. First, the AST_REDIRECTING_REASON_SEND_TO_VM value had to be defined in callerid.h, and its string value and definition had to be added to a table in callerid.c. Finally, parsing of the Diversion header had to be added to handle_request_refer() in chan_sip.c
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> Diffs
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>   /certified/branches/1.8.11/channels/chan_sip.c 366547 
>   /certified/branches/1.8.11/include/asterisk/callerid.h 366547 
>   /certified/branches/1.8.11/main/callerid.c 366547 
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> Diff: https://reviewboard.asterisk.org/r/1925/diff
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> Testing
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> Testing is posted as a separate review: https://reviewboard.asterisk.org/r/1926
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> Thanks,
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> Mark
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