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

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233580#comment-233580 ] 

Rusty Newton commented on ASTERISK-26569:
-----------------------------------------

Since the issue is now the particular behavior with one specific softphone during an ARI redirect , can you open a new issue for that report, and post the debug there? That will make it easier for those coming to issue to decipher what we are investigating.

However, your log file didn't include the DEBUG or verbose options. If you open a new issue please attach a log of the problem that includes "warning,notice,error,verbose,debug" and be sure that verbose and debug are both turned up to level 5 or above.

I'll go ahead and close this issue out since the original issue was not a problem.

> 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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