[asterisk-bugs] [JIRA] (ASTERISK-21673) Asterisk conversion of 183 without SDP to 180 breaks interoperability with Microsoft Lync

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21673?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205874#comment-205874 ] 

Rusty Newton commented on ASTERISK-21673:
-----------------------------------------

The consensus here is that this is not a bug, but essentially a feature request for interoperability functionality between Asterisk and Microsoft Lync (possibly other systems unamed). 

Since we don't generally accept feature requests without a patch I'm going to close this out as Suspended until someone provides a patch. 

As you mentioned this feature would not be released in Asterisk until Asterisk 12 (that does not necessarily mean that the patch wouldn't be compatible with 11, if you wanted to run a backported version..) Here's more information on our [New Feature Process|https://wiki.asterisk.org/wiki/display/AST/New+Feature+Process].

For further discussion and if you want to try to find someone to write a patch for you - I recommend bringing this feature request up on the asterisk-users and asterisk-dev lists. asterisk-dev is probably the best place to find a dev who may be interested. Some members of the community even offer [bounties|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties] on their bugs to increase the incentive for getting help.

If you or someone else starts working on the issue and provides a patch you can feel free to comment on here, the asterisk-dev list or the #asterisk-bugs IRC chatroom to have the issue re-opened.
                
> Asterisk conversion of 183 without SDP to 180 breaks interoperability with Microsoft Lync
> -----------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-21673
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21673
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 11.3.0
>         Environment: FreePBX distro 3.211.63-7
> Centos 6.3, Asterisk 11.3.0 
>            Reporter: dcitelecom
>         Attachments: Early media Asterisk 11 log.txt
>
>
> Voxbone DID (defined in inbound routes) is sent out to customer trunk ABCco (TCP) trunk. They reply with 183 session progress (early media announcement) and Asterisk sends 180 ringing back to Voxbone.
> This is problematic where the DID is forwarded to a cell phone and the phone is off or the forwarding number was entered wrong. In both cases the user should hear an early media announcement that the user is not available or the number is not in service. Instead they hear a ring tone and don't know there is a problem. Log attached
> {noformat}
> DIALPLAN
> exten => #0216479999999,1,Set(__FROM_DID=$
> {EXTEN}
> )
> exten => #0216479999999,n,Gosub(app-blacklist-check,s,1())
> exten => #0216479999999,n,Gosub(sub-record-cancel,s,1())
> exten => #0216479999999,n,Set(__REC_POLICY_MODE=never)
> exten => #0216479999999,n,Set(CDR(did)=$
> {FROM_DID}
> )
> exten => #0216479999999,n,ExecIf($[ "$
> {CALLERID(name)}
> " = "" ] ?Set(CALLERID(name)=$
> {CALLERID(num)}
> ))
> exten => #0216479999999,n,Set(CHANNEL(musicclass)=none)
> exten => #0216479999999,n,Set(__MOHCLASS=none)
> exten => #0216479999999,n,Set(__CALLINGPRES_SV=$
> {CALLERPRES()}
> )
> exten => #0216479999999,n,Set(CALLERPRES()=allowed_not_screened)
> exten => #0216479999999,n(dest-ext),Goto(ext-trunk,5,1)
> {noformat}
> {noformat}
> [ABCAsia] (trunk 5)
> disallow=all
> type=peer
> host=202.xxx.xxx.xxx
> transport=tcp
> allow=alaw
> allow=ulaw
> relaxdtmf=yes
> dtmfmode=rfc2833
> directmedia=yes
> context=from-ABCco-HK
> {noformat}

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