<p>Hi<br>
Try Nat=yes in general settings </p>
<p>On 06-Nov-2010 9:57 PM, "Silver Thorne" <<a href="mailto:zoraxus@gmail.com">zoraxus@gmail.com</a>> wrote:<br type="attribution">> Let me explain:<br>> <br>> When I dial into Asterisk ( I have a SIP trunk - which I need to make <br>
> sure is not faulty), I only get one-way voice communication.<br>> The calling party, from the SIP trunk hears nothing - the extension <br>> rings on the Asterisk server (you can see it in the CLI and hear it at <br>
> the computer), and the softphone rings<br>> <br>> However, when you answer the SIP softphone , you can only hear the voice <br>> FROM the softphone out.<br>> <br>> Where would I start to troubleshoot this? I am a little clueless!<br>
> <br>> Thanks for all of your help.<br>> <br>> Asterisk 1.4.31 built by root @ <a href="http://some_server.foo.net">some_server.foo.net</a> on a x86_64 running <br>> Linux on 2010-06-10 14:32:34 UTC<br>> <br>
> Sip Settings:<br>> <br>> Global Settings:<br>> ----------------<br>> SIP Port: 5060<br>> Bindaddress: 0.0.0.0<br>> Videosupport: No<br>> AutoCreatePeer: No<br>
> Allow unknown access: Yes<br>> Allow subscriptions: Yes<br>> Allow overlap dialing: Yes<br>> Promsic. redir: No<br>> SIP domain support: No<br>> Call to non-local dom.: Yes<br>
> URI user is phone no: No<br>> Our auth realm asterisk<br>> Realm. auth: No<br>> Always auth rejects: No<br>> Call limit peers only: No<br>> Direct RTP setup: No<br>
> User Agent: Asterisk PBX<br>> MWI checking interval: 10 secs<br>> Reg. context: (not set)<br>> Caller ID: asterisk<br>> From: Domain:<br>> Record SIP history: Off<br>
> Call Events: Off<br>> IP ToS SIP: none<br>> IP ToS RTP audio: none<br>> IP ToS RTP video: none<br>> T38 fax pt UDPTL: No<br>> RFC2833 Compensation: No<br>
> SIP realtime: Disabled<br>> <br>> Global Signalling Settings:<br>> ---------------------------<br>> Codecs: 0x8000e (gsm|ulaw|alaw|h263)<br>> Codec Order: none<br>
> T1 minimum: 100<br>> No premature media: No<br>> Relax DTMF: No<br>> Compact SIP headers: No<br>> RTP Keepalive: 0 (Disabled)<br>> RTP Timeout: 0 (Disabled)<br>
> RTP Hold Timeout: 0 (Disabled)<br>> MWI NOTIFY mime type: application/simple-message-summary<br>> DNS SRV lookup: Yes<br>> Pedantic SIP support: No<br>> Reg. min duration 60 secs<br>
> Reg. max duration: 3600 secs<br>> Reg. default duration: 120 secs<br>> Outbound reg. timeout: 20 secs<br>> Outbound reg. attempts: 0<br>> Notify ringing state: Yes<br>> Notify hold state: No<br>
> SIP Transfer mode: open<br>> Max Call Bitrate: 384 kbps<br>> Auto-Framing: No<br>> <br>> Default Settings:<br>> -----------------<br>> Context: default<br>
> Nat: RFC3581<br>> DTMF: rfc2833<br>> Qualify: 0<br>> Use ClientCode: No<br>> Progress inband: Never<br>> Language: (Defaults to English)<br>
> MOH Interpret: default<br>> MOH Suggest:<br>> Voice Mail Extension: asterisk<br>> <br>> ----<br>> Parsing /etc/asterisk/extconfig.conf<br>> <br>> sip show peer<br>> <br>> * Name : 155<br>
> Secret :<Set><br>> MD5Secret :<Not set><br>> Context : extern<br>> Language : en<br>> AMA flags : Unknown<br>> Transfer mode: open<br>> MaxCallBR : 384 kbps<br>
> CallingPres : Presentation Allowed, Not Screened<br>> Call limit : 0<br>> Callgroup :<br>> Pickupgroup :<br>> Callerid : "Glen's Sysadmin Test Line"<200111222><br>
> ACL : No<br>> Codec Order : (none)<br>> Auto-Framing: No<br>> <br>> <br>> <br>> -- <br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></p>