[asterisk-bugs] [JIRA] (ASTERISK-20282) Call pickup incompatibility with Cisco 1760V + RPID

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20282?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=198822#comment-198822 ] 

Matt Jordan commented on ASTERISK-20282:
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{quote}
in the interim, is the ip address in the rpid field correct?
{quote}

Yes, Asterisk should be providing the Remote-Party-ID of the called party to the calling party.  This allows the calling party to make decisions about the remote called party (see this [draft|http://tools.ietf.org/html/draft-ietf-sip-privacy-04] for reference - as it turns out, Remote-Party-ID is not an RFC).

In your particular case, Asterisk is sending a re-INVITE followed by an UPDATE request to 9910 at 192.168.1.7 when the call pickup is initiated, as this is the first time it "knows" who the remote party should be.  The problem here is that Asterisk initially sends a Remote-Party-ID of {{"Sales Desk 2" <sip:112 at 192.168.1.7>;party=called;privacy=off;screen=no}} (re-INVITE) followed by {{"testing" <sip:139 at 192.168.1.7>;party=calling;privacy=off;screen=no}} (UPDATE).  It appears as if it is the UPDATE request that throws the Cisco off.

The UPDATE request here is interesting, as I'm not sure that Asterisk should be sending that one.  Its possible that something weird is happening with a combination of connected line updates; I'll have to dig into that it a bit more.  I'll reopen this issue for now.
                
> Call pickup incompatibility with Cisco 1760V + RPID
> ---------------------------------------------------
>
>                 Key: ASTERISK-20282
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20282
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/Interoperability
>    Affects Versions: 1.8.15.0
>            Reporter: Jeremy Kister
>            Assignee: Rusty Newton
>         Attachments: cisco-group-with-rpid.txt, extensions.conf, features.conf, group-without-rpid.txt, group-with-rpid.txt, single-with-rpid.txt, sip.conf
>
>
> I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to 
> asterisk 1.8.15.0.
> imagining in extensions.conf:
> exten => 1,1,Dial(SIP/121)
> exten => 2,1,Dial(SIP/121&SIP/122)
> When a caller dials extension 2 /and/ I have
>   trustrpid=yes
>   generaterpid=yes
>   sendrpid=yes
> in sip.conf and I use the pickup exten, the caller is disconnected.
> if i set the rpid generate/send = no for the cisco peer, the user is 
> connected.
> calls to exten 1 work regardless of rpid settings.

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