[asterisk-bugs] [JIRA] (ASTERISK-28739) Dropping redundant connected line update

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28739?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team updated ASTERISK-28739:
-------------------------------------

    Status: Waiting for Feedback  (was: Waiting for Feedback)

> Dropping redundant connected line update
> ----------------------------------------
>
>                 Key: ASTERISK-28739
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28739
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/General
>    Affects Versions: 13.29.2
>         Environment: Debian 9 64 bit
>            Reporter: David Cunningham
>            Assignee: Asterisk Team
>            Severity: Minor
>
> We are seeing a "Dropping redundant connected line update" message in the Asterisk log which appears to be incorrect.
> The channel with the update is created by a Dial out from Asterisk to a SIP telephone:
> [Feb 13 14:05:01] DEBUG[2899] manager.c: Examining AMI event:
> Event: Newexten
> Privilege: call,all
> Timestamp: 1581563101.208429
> Channel: SIP/AAA.AAA.52.245:5060-000002f0
> ChannelState: 0
> ChannelStateDesc: Down
> CallerIDNum: 897
> CallerIDName: John
> ConnectedLineNum: 897
> ConnectedLineName: John
> Language: en
> AccountCode: 
> Context: from-external
> Exten: 171373917
> Priority: 1
> Uniqueid: 1581563101.2732
> Linkedid: 1581563101.2729
> Extension: 171373917
> Application: AppDial
> AppData: (Outgoing Line)
> This results in an INVITE:
> [Feb 13 14:05:01] VERBOSE[23326][C-0000019a] chan_sip.c: Reliably Transmitting (NAT) to AAA.AAA.52.245:5060:
> INVITE sip:171373917 at AAA.AAA.52.245:5060 SIP/2.0
> Via: SIP/2.0/UDP AAA.AAA.52.245:5070;branch=z9hG4bK76f896c2;rport
> Max-Forwards: 70
> From: "John" <sip:897 at AAA.AAA.52.245:5070>;tag=as0795e872
> To: <sip:171373917 at AAA.AAA.52.245:5060>
> Contact: <sip:897 at AAA.AAA.52.245:5070>
> Call-ID: 2baf801c5631714d53c92b27735249a6 at AAA.AAA.52.245:5070
> Later we try to update the callerid to 0123456789 via the AMI however Asterisk rejects it:
> [Feb 13 14:05:07] DEBUG[23424] manager.c: Running action 'Login'
> [Feb 13 14:05:07] DEBUG[23424] manager.c: Running action 'Setvar'
> [Feb 13 14:05:07] DEBUG[23424] channel.c: SIP/AAA.AAA.52.245:5060-000002f0: Dropping redundant connected line update "0123456789" <0123456789>.
> [Feb 13 14:05:07] DEBUG[23424] manager.c: Running action 'Logoff'
> Reading the Asterisk code, I believe it should only be considered redundant if the callerid is unchanged, but here it really is different.
> Please let us know what else is needed to debug this issue. Thank you.



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