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

[Daniele Pallastrelli (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233572#comment-233572 ] 

Daniele Pallastrelli commented on ASTERISK-26569:
-------------------------------------------------

I attached the console output for the flooding (BTW: I erased from the file 470000 lines containing the same warning message :-) ) 
I had to redirect the stdout to a file.

As for my application: I understand that the REFER request can work with some endpoint and not with other. Maybe Yealink phone handle the request int he right way while PhonerLite don't.
Anyway I'll go for the solution you suggested: I'll create a bridge and an outgoing channel. Moreover, I prefer to mantain the control of the channels, so I guess this should be my first option.
Thanks a lot.

> 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
>            Severity: Minor
>         Attachments: console_26569.txt, debug_log_26569, output_26569.txt
>
>
> 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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