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

George Joseph (JIRA) noreply at issues.asterisk.org
Mon Feb 17 11:15:25 CST 2020


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

George Joseph updated ASTERISK-28739:
-------------------------------------

    Assignee: David Cunningham  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

It's still a little unclear what the scenario and your objective are.

Setting CONNECTEDLINE on an outgoing channel is going to go back into the Asterisk core to the incoming channel.  Is that what you wanted?

See this wiki page fore mote info on CALLERID and CONNECTEDLINE.
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information




> 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: David Cunningham
>            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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