[asterisk-bugs] [JIRA] (ASTERISK-28983) Unable to redirect outgoing calls to mobile

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28983?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=251376#comment-251376 ] 

Joshua C. Colp commented on ASTERISK-28983:
-------------------------------------------

We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines



> Unable to redirect outgoing calls to mobile
> -------------------------------------------
>
>                 Key: ASTERISK-28983
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28983
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 17.0.0
>            Reporter: Rozi Wagency
>
> Hi, 
> I have been taking care of the Asterisk server for a few weeks.
> Before my arrival my predecessor had installed Asterisk.
> The server uses SIP and E1 communication
> Today, with the COVID-19, our users need to forward their call from their landline to a mobile phone.
> This function does not work. When the user makes a return to his mobile from the Landline phone, this does not work.
> However, when you indicate hard on the Asterisk server, it works.
> My manager does not want to indicate in hard on the server, she absolutely wants to let the users to indicate their number freely. I looked but I couldn't find the problem. Could you help me please.
> below the error message.
> "== Using SIP RTP CoS mark 5
> [Jul  1 12:01:46] WARNING[8395][C-0000001f]: app_dial.c:2432 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)"



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