asterisk users
2013-May-21 17:11 UTC
[asterisk-users] Failed to authenticate device "Ext 110"
I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" < sip:110 at 192.168.6.2>;tag=1130259112 Asterisk 192.168.6.2 OpenVPN on router 10.8.0.1 Remote Yealink phone 10.8.0.6 The remote phone shows as being registered: PBX*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN Also, if there is voicemail in the mailbox for 110, the phone's message light is lit and it beeps periodically. toshi*CLI> sip show peer 110 * Name : 110 Description : Yealink OpenVPN Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : remote-phones Record On feature : automon Record Off feature : automon Subscr.Cont. : <Not set> Language : Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : 110 VM Extension : asterisk LastMsgsSent : 1/0 Call limit : 4 Max forwards : 0 Dynamic : Yes Callerid : "Ext 110" <110> MaxCallBR : 384 kbps Expire : 608 Insecure : no Force rport : No Symmetric RTP: No ACL : Yes DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : Yes Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 10.8.0.6:5062 Defaddr->IP : 10.8.0.6:5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 110 SIP Options : (none) Codecs : (ulaw) Codec Order : (ulaw:20) Auto-Framing : No Status : OK (237 ms) Useragent : Yealink SIP-T28P 2.61.23.3 00:15:65:xx.xx.xx Reg. Contact : sip:110 at 10.8.0.6:5062 Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No sip.conf: [110] context=remote-phones type=peer host=dynamic qualify=1500 canreinvite=no dtmfmode=rfc2833 progressinband=no callgroup=1 pickupgroup=1 ; We can do call pickup for call group 1 call-limit=4 busy-level=1 qualify=yes deny=0.0.0.0/0.0.0.0 permit=0.0.0.0/0.0.0.0 nat=no qualify=8000 description=Yealink OpenVPN defaultuser=110 secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid="Ext 110" <110> mailbox=110 defaultip=10.8.0.6 port=5060 disallow=all allow=ulaw Any suggestions on what might be happening here, and how it could be resolved? THANKS ALL! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130521/d6726852/attachment.htm>
Matthew J. Roth
2013-May-21 17:26 UTC
[asterisk-users] Failed to authenticate device "Ext 110"
asterisk users wrote:> > I'm having a strange problem recently with a Yealink SIP-T28P phone connected > to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I > dial anything from the phone, it shows "Forbidden", and the Asterisk console > shows: > > [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" < sip:110 at 192.168.6.2 >;tag=1130259112 > > Asterisk 192.168.6.2 > OpenVPN on router 10.8.0.1 > Remote Yealink phone 10.8.0.6 > > The remote phone shows as being registered: > PBX*CLI> sip show peers > Name/username Host Dyn Forcerport ACL Port Status Description > 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN > > Also, if there is voicemail in the mailbox for 110, the phone's message light > is lit and it beeps periodically. > > ... > > Any suggestions on what might be happening here, and how it could be resolved?That is quite strange. Please provide SIP traces of the dialogs between Asterisk and the phone in the following two scenarios: 1) Phone registering to Asterisk (presumably successful) 2) Phone dialing to Asterisk (presumably unsuccessful) Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer