[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=233538#comment-233538 ] 

Joshua Colp commented on ASTERISK-26569:
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So the REFER is telling the softphone to contact "toronto/101 at 192.168.210.132". If you replace SIP/toronto/101 with SIP/101 at toronto it may work better. This does mean, though, that the softphone will no longer be connected to the system where it entered the ARI application and if any authentication is required by the other server (toronto) your softphone may not be able to authenticate.

> 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: console_26569.txt, 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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