[asterisk-bugs] [JIRA] Commented: (ASTERISK-16875) chan_sip returns 503 Service Unavailable response, when additional information would allow it to return 480 response

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-16875?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197512#comment-197512 ] 

Matt Jordan commented on ASTERISK-16875:
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Asterisk is a B2BUA.  As such, correct configuration in the dialplan would allow the administrator to specify the desired hangup cause.  The fact that Congestion was specified from the dialplan indicates that something other than the desired endpoint not being available occurred - either because someone indicated Congestion to the dialing UA, or because Congestion actually did occur on the other channel technology.  Either way, this could be intercepted and handled without relying on hard coded logic that is - in many circumstances - potentially incorrect.

Also, your patch falls through the various case statements, so this patch will definitely result in incorrect behavior.

Because the correct way of handling this in Asterisk would be to set the desired hangup cause in the Hangup application, this issue will be closed out as "Not a Bug".

> chan_sip returns 503 Service Unavailable response, when additional information would allow it to return 480 response
> --------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-16875
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-16875
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/Interoperability
>            Reporter: Szabolcs
>            Severity: Minor
>         Attachments: actualdebug, channel.c.patch, chan_sip_20101105.c.patch, chan_sip.c.patch
>
>
> When a phone not responding (HangupCauseCode:20) the caller from the sip trunk receive SIP response 503. The correct SIP response "Temporarily unavailable" 480!  (Also occur Asterisk 1.6.2.12)
> ****** ADDITIONAL INFORMATION ******
> SIP/2.0 503 Service Unavailable
> Via: SIP/2.0/UDP 10.16.91.2:5060;branch=z9hG4bK-d8754z-2f6c0a4862167d1a-1---d8754z-;received=10.16.91.2;rport=5060
> From: <sip:anonymous at anonymous.invalid>;tag=92057525eb23103341d0d007ef3f9464
> To: <sip:+36xxxxx at 10.16.91.3>;tag=as16035486
> Call-ID: 0000115d4cc724c77af1ea49f0f0f0f0 at 10.35.1.1/1
> CSeq: 101 INVITE
> Server: Asterisk PBX 1.8.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> X-Asterisk-HangupCause: Unknown
> X-Asterisk-HangupCauseCode: 20
> Content-Length: 0

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