Let me explain: When I dial into Asterisk ( I have a SIP trunk - which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing - the extension rings on the Asterisk server (you can see it in the CLI and hear it at the computer), and the softphone rings
However, when you answer the SIP softphone , you can only hear the voice FROM the softphone out. Where would I start to troubleshoot this? I am a little clueless! Thanks for all of your help. 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 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 -- _____________________________________________________________________ -- 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
