Steve; You are so right - it was the end of the day, I was tired and pissy.
Let me try this again: Version: ns211156*CLI> core show version Asterisk 1.4.31 built by root @ some_server.foo.net on a x86_64 running Linux on 2010-06-10 14:32:34 UTC Name and version of endpoints involved: Sip Settings: Global Settings: ---------------- SIP Port: 5060 Bindaddress: 0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: Yes Promsic. redir: No SIP domain support: No Call to non-local dom.: Yes URI user is phone no: No Our auth realm asterisk Realm. auth: No Always auth rejects: No Call limit peers only: No Direct RTP setup: No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off IP ToS SIP: none IP ToS RTP audio: none IP ToS RTP video: none T38 fax pt UDPTL: No RFC2833 Compensation: No SIP realtime: Disabled Global Signalling Settings: --------------------------- Codecs: 0x8000e (gsm|ulaw|alaw|h263) Codec Order: none T1 minimum: 100 No premature media: No Relax DTMF: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Default Settings: ----------------- Context: default Nat: RFC3581 DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: (Defaults to English) MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk ---- Parsing /etc/asterisk/extconfig.conf sip show peer * Name : 155 Secret :<Set> MD5Secret :<Not set> Context : extern Language : en AMA flags : Unknown Transfer mode: open MaxCallBR : 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup : Pickupgroup : Callerid : "Glen's Sysadmin Test Line"<200111222> ACL : No Codec Order : (none) Auto-Framing: No sip.conf [general] ;port = 5060 ;bindaddr=0.0.0.0 ;srvlookup=yes ;context=extern ;nat=yes ;localnet=192.168.0.0/255.255.0.0 ;allowguest=no [Axialys] type=peer host=sip-proxy.xxx.xxx.net fromuser=USERID_1 secret=password-1 qualify=yes context=extern quality=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw nat=yes insecure=port,invite [Axialys2] type=peer host=sip-proxy.xxx.xxx.net host=dynamic fromuser=userid_1 secret=password_1 qualify=yes context=extern quality=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw nat=yes insecure=port,invite [GlenAxialys3] type=peer host=sip-proxy.xxx.xxx.net fromuser=userid_1 secret=password_1 qualify=yes context=extern quality=yes dtmfmode=rfc2833 disallow=all allow=gsm allow=ulaw nat=yes insecure=port,invite [Nov 2 17:10:04] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:06] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:07] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:09] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:09] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:17] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:24] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:24] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:31] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:31] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. [Nov 2 17:10:35] NOTICE[13804] chan_sip.c: Call from '6839' to extension '33173793697' rejected because extension not found. So, when I call the 33173793697 number, the above entry is what I see in the log. Glen On 11/1/2010 17:32, Steve Edwards wrote: > On Mon, 1 Nov 2010, Silver Thorne wrote: > >> > Anyone see this before: >> > >> > [Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have >> > <6839>, digest has<3169> > You may have better luck with a more descriptive subject. Lots of users > have an issue or two with Asterisk. > > Some details will also help. Like: > > ) Version of Asterisk. > ) Name and version of the endpoints involved. > > ) Relevant sections of sip.conf as well as the console output from 'sip > show settings,' 'sip show user<username>,' and 'sip show peer > <peername>.' (I'm a 1.2 Luddite.) > > ) Console output of 'sip debug ip<address>' illustrating the 'issue.' > > Don't forget to 'sanitize' any IP addresses, usernames, and passwords that > you consider valuable. (Actually, it would be better to redo your > configuration with 'throw-away' credentials (like username1 and password1) > for the duration of your issue -- less chance of exposing something or > mistyping an important detail.) > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
