[asterisk-bugs] [JIRA] (ASTERISK-28214) PJSIP and CHAN_SIP issues
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Dec 17 12:39:48 CST 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-28214?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team closed ASTERISK-28214.
------------------------------------
> 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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