[asterisk-dev] 3PCC patch for AMI "SIPnotify"

[Matt Fredrickson]

Answer below.

On Thu, Nov 30, 2017 at 6:46 PM, Yasuhiko Kamata
<yasuhiko.kamata at nxtg.co.jp> wrote:
> Hello asterisk-dev list,
>
> We have created a patch for use in 3PCC applications.
> With this patch, asterisk can let the certain SIP phone answer or hold
> through AMI action.
>
> [summary]
> A patch for sending in-dialog SIP NOTIFY message
> with "SIPnotify" AMI action for latest (14.x and 15.x) asterisk.
>
> [detail]
> asterisk has an AMI action called "SIPnotify" in order to send any SIP
> NOTIFY message. We want to let the SIP phone answer or hold by using
> this action, but some SIP phones, especially AudioCodes 4xx phone,
> do not accept it.
>
> According to our investigation, some SIP phones do not accept SIP
> NOTIFY message if it's not an in-dialog (must have the same "From:",
> "Call-ID", and tags as "INVITE"), so we created this patch.
>
> [note]
> Some additional (not default) settings may be required on the phone side.
> For AudioCodes' 4xx phone, these settings should be added:
>
> voip/talk_event/enabled=1
> voip/auto_answer/enabled=1
> voip/auto_answer_use_180/enabled=1
>
> [how to use]
> After applying this patch (and rebuild after that),
> "SIPnotify" can be sent just like this:
>
> ---
> Action: SIPnotify
> ActionID: 3
> Channel: SIP/1001
> Variable: Event=talk
> Variable: Call-ID=112233445566778899aabbccddeeff at example.org:5060
> ---
>
> Here, channel and SIP call-ID should be changed accordingly.
> SIP call-ID can be acquired by "SIPCALLID" in response of
> "Status" AMI action:
>
> ---
> Action: Status
> ActionID: 2
> Channel: SIP/1001-00000001
> AllVariables: true
>
> Event: Status
> Privilege: Call
> Channel: SIP/1001-00000001
> ChannelState: 5
> ChannelStateDesc: Ringing
> ...
> Variable: DIALEDPEERNUMBER=1001
> Variable: SIPCALLID=112233445566778899aabbccddeeff at example.org:5060
> ---
>
> After sending a "SIPnotify" action with "Variable: Call-ID=...",
> SIP NOTIFY message will be sent to the target SIP phone with in-dialog
> (i.e. the same "From:", "Call-ID", and tags as "INVITE").
>
> If change "Event=talk" to "Event=hold", SIPnotify can be used to let
> the phone hold (use "Event=talk" to unhold).
>
> Thanks,
>
> --
> Yasuhiko Kamata <yasuhiko.kamata at nxtg.co.jp>

Hey Yasuhiko,

First off, thanks for letting us know about your interesting patch!
The best way to get a patch contributed is to submit it to our code
review site, gerrit.asterisk.org.  From there, members of the Asterisk
development team can review the code and further discuss it in a more
appropriate context for code review.

You can find some information about the patch contribution process at
https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process

Best wishes.

-- 
Matthew Fredrickson
Digium, Inc. | Asterisk Project Lead and Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA