[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=233565#comment-233565 ] 

Joshua Colp edited comment on ASTERISK-26569 at 11/10/16 9:44 AM:
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Doing so shouldn't result in that flood so I'd suggest uploading a console output of that.

As for it not working - this goes back to one of my earlier comments. We are telling the phone to go elsewhere and it is up to it to do that. Not all implementations are created equal and not all expect it to occur or handle it properly. From the Asterisk side there's nothing we can do to change that. Your other option as previously mentioned is to create an outgoing channel in Asterisk and bridge them. That will work all the time.


was (Author: jcolp):
Doing so shouldn't result in that flood so I'd suggest uploading a console output of that.

As for it not working - this goes back to one of my earlier comments. We are telling the phone to go elsewhere and it is up to it to do that. Not all implementations are created and not all expect it to occur or handle it properly. From the Asterisk side there's nothing we can do to change that. Your other option as previously mentioned is to create an outgoing channel in Asterisk and bridge them. That will work all the time.

> 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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