[asterisk-bugs] [JIRA] (ASTERISK-28214) PJSIP and CHAN_SIP issues

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28214?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=245775#comment-245775 ] 

Asterisk Team commented on ASTERISK-28214:
------------------------------------------

We appreciate the difficulties you are facing, however information request type issues 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.

If this issue is actually a bug please use the Bug issue type instead.

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

> PJSIP and CHAN_SIP issues
> -------------------------
>
>                 Key: ASTERISK-28214
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28214
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 13.22.0
>         Environment: FreePBX 14 
> Asterisk 13.22.0 c
>            Reporter: Anthony Kay
>              Labels: pjsip
>
> I used to setup FreePBX pre version 12 where there was no pjsip and chansip. 
> Upon setting them up in version 14 I am having issues with PJSIP and SIP.....
> I read an article to set FreePBX to use only chan sip which I have set within the advanced settings - I watched a video on how to do this, and then it said that you set nat as yes, then within asterisk sip settings you set the chan sip bind port to 5060. I have successfully registerest my endpoint and my trunks, however when trying to call a number I continuously get 
> =========================================================================
> Connected to Asterisk 13.22.0 currently running on IT-FE-PBX-01 (pid = 2495)
> [2018-12-17 20:25:43] WARNING[2598][C-00000006]: chan_sip.c:17266 check_auth: username mismatch, have <1234>, digest has <IP Phone>
> [2018-12-17 20:25:43] NOTICE[2598][C-00000006]: chan_sip.c:26364 handle_request_invite: Failed to authenticate device <sip:10.17.241.100>;tag=1629616225
> [2018-12-17 20:33:28] WARNING[2598][C-00000007]: chan_sip.c:17266 check_auth: username mismatch, have <1234>, digest has <IP Phone>
> [2018-12-17 20:33:28] NOTICE[2598][C-00000007]: chan_sip.c:26364 handle_request_invite: Failed to authenticate device <sip:10.17.241.100>;tag=3369418603
> The endpoint is registered to my pbx but when dialling out I keep getting the above error when runnning asterisk -rvvvvv



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