Hi Try Nat=yes in general settings On 06-Nov-2010 9:57 PM, "Silver Thorne" <[email protected]> wrote: > 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
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