[asterisk-bugs] [JIRA] (ASTERISK-18999) Connected Line updates need a disable option on a per SIP device / trunk basis.

[Richard Mudgett (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-18999?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Richard Mudgett closed ASTERISK-18999.
--------------------------------------

    Resolution: Not A Bug

Based upon [~mmichelson]'s analysis this is not a bug but bad dialplan and misbehavior of the telco as a result.

> Connected Line updates need a disable option on a per SIP device / trunk basis.
> -------------------------------------------------------------------------------
>
>                 Key: ASTERISK-18999
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18999
>             Project: Asterisk
>          Issue Type: New Feature
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.7.1, 13.18.4
>         Environment: CentOS 5.5 and 6.0, FreePBX 2.9
>            Reporter: Gary Herbstman
>            Severity: Minor
>              Labels: asterisk, colp
>         Attachments: Asterisk capture dropped call with sip debug.txt, cap-cell-ivr-rg-ext7999-vp-filtered.pcap, Failed call flow.jpg
>
>
> Some SIP trunk providers do not handle the connected line updates well and are dropping calls. This feature can be disabled by setting senderpid=no on the trunk but this also prevents CallerID / Remote-PartyID to be set on the initial INVITE as well. We need to be abble to set the CallerID on the initial INVITE but NOT send additional INVITEs to the trunk as the call is transferred.
> The trunk provider Voicepulse in this case but seems to also occur with CallCentric, is sending back ERROR 491 and dropping the call.
> This is somewhat related to this DAHDI bug ASTERISK-17025. Same problem but different code.
> As a further note, Asterisk is sending not one but two updates to the trunk. One for device at extension and one for user name at extension.
>  



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