[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=245776#comment-245776 ] 

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

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 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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