[asterisk-bugs] [JIRA] (ASTERISK-26569) ari: Redirect does not work over sip trunk

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233531#comment-233531 ] 

Joshua Colp commented on ASTERISK-26569:
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ARI provides the building blocks to write that. It does not provide a specific operation to do it, as that's really application logic. Using a POST to channels[1] you can create an outgoing channel to where you want. Once answered you can create a bridge[2] and add both channels[3] to allow them to talk.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Channels+REST+API#Asterisk14ChannelsRESTAPI-originate
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Bridges+REST+API#Asterisk14BridgesRESTAPI-create
[3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Bridges+REST+API#Asterisk14BridgesRESTAPI-addChannel

> ari: Redirect does not work over sip trunk
> ------------------------------------------
>
>                 Key: ASTERISK-26569
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26569
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/Stasis, Resources/res_ari_channels
>    Affects Versions: 13.10.0, 14.0.2, GIT
>         Environment: Linux debian-8 3.16.0-4-586 #1 Debian 3.16.36-1+deb8u1 (2016-09-03) i686 GNU/Linux
> gcc (Debian 4.9.2-10) 4.9.2
>            Reporter: Daniele Pallastrelli
>            Assignee: Unassigned
>         Attachments: debug_log_26569
>
>
> h4. Frequency
> Systematic issue.
> h4. Symptoms
> ARI redirect channel over sip trunk does not work.
> h4. Steps required to reproduce the issue
> With the following setup:
> * extension 6100 that answer and then put the channel in a stasis app called "attendant"
> * a phone named 290
> * a sip trunk called "toronto" towards another asterisk installation with a phone named 101
> A phone (290) calls the extension 6100 (stasis application), resulting in a channel with id:
>   pc_dany_asterisk-1478699776.2
> Then, try to redirect the channel towards the phone 101 over the trunk "toronto":
> {code}
> curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels/pc_dany_asterisk-1478699776.2/redirect?endpoint=sip/toronto/101"
> {code}
> h4. Expected Behaviour
> After the http POST, the local phone should result connected to the remote phone (according to https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Channels+REST+API#Asterisk14ChannelsRESTAPI-redirect)
> h4. Behaviour actually encountered
> The redirect operation fails with the message:
> {{handle_request_invite: Call from '290' (192.168.210.111:5060) to extension 'toronto101' rejected because extension not found in context 'LocalSets'.}}
> (logs attached)
> h4. Notes
> It seems there is a problem parsing the "endpoint" parameter in the request. The slash between "toronto" and "101" is missing (in the request I send "sip/toronto/101" but in the log message I get "toronto101")
> Other http requests (e.g., channel creation) works with sip trunks. The url of the request has the same syntax as for the endpoint parameter:
> {code}
> POST http://192.168.210.132:8088/ari/channels?endpoint=sip/toronto/101&app=attendant
> {code}
> I tested the bug also with the last (at the time of writing this) commit in the git repository: GIT-master-0d85f18



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