<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body text="#000000" bgcolor="#ffffff">
<br>
Hello All;<br>
<br>
I have more clues that may assist in resolving this:<br>
<br>
If I use the same softphone and dial out with the same Asterisk
server. The SIP/voice traffic is able to be heard in both
directions.<br>
<br>
So, anyone have any ideas for me? Still a little clueless.<br>
<br>
Glen<br>
On 11/6/2010 13:00, Zuhair Raza wrote:
<blockquote
cite="mid:AANLkTin4VsWPR0G_oq0PzxN16ZnbmBjkP3HFEEPfrhj3@mail.gmail.com"
type="cite">
<p>Hi<br>
Try Nat=yes in general settings </p>
<p>On 06-Nov-2010 9:57 PM, "Silver Thorne" <<a
moz-do-not-send="true" 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 moz-do-not-send="true"
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
moz-do-not-send="true" 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 moz-do-not-send="true"
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 moz-do-not-send="true"
href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</p>
</blockquote>
</body>
</html>