<div>thanks Jim</div>
<div> </div>
<div>I will check stun server settings asap, </div>
<div> </div>
<div>but i have noticed 192.168.x.x is also present in the debug of successful call having both way audio. so i don't think this has to do anything with this.</div>
<div> </div>
<div>below is the sip debug of successful call .</div>
<div> </div>
<div>---</div>
<div><span lang="EN">
<p>Audio is at 79.80.154.99 port 14034</p>
<p>Adding codec 0x8 (alaw) to SDP</p>
<p>Adding codec 0x4 (ulaw) to SDP</p>
<p>Adding codec 0x2 (gsm) to SDP</p>
<p>Adding non-codec 0x1 (telephone-event) to SDP</p>
<p>Reliably Transmitting (NAT) to <a href="http://116.18.35.235:28614">116.18.35.235:28614</a>:</p>
<p>INVITE sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport</p>
<p>From: "pepsi coke" <<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>>;tag=as12245807</p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588></p>
<p>Contact: <<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>></p>
<p>Call-ID: <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>CSeq: 102 INVITE</p>
<p>User-Agent: Asterisk PBX</p>
<p>Max-Forwards: 70</p>
<p>Date: Wed, 21 Jul 2010 15:06:24 GMT</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</p>
<p>Supported: replaces</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 285</p>
<p>v=0</p>
<p>o=root 9626 9626 IN IP4 79.80.154.99</p>
<p>s=session</p>
<p>c=IN IP4 79.80.154.99</p>
<p>t=0 0</p>
<p>m=audio 14034 RTP/AVP 8 0 3 101</p>
<p>a=rtpmap:8 PCMA/8000</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:3 GSM/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=silenceSupp:off - - - -</p>
<p>a=ptime:20</p>
<p>a=sendrecv</p>
<p>---</p>
<p>[Jul 21 11:06:24] WARNING[23749]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms.</p>
<p>-- Called adf</p>
<p>ast-server*CLI> </p>
<p><--- SIP read from <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p>SIP/2.0 180 Ringing</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678</p>
<p>Contact: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588></p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350</p>
<p>From: "pepsi coke"<<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>>;tag=as12245807</p>
<p>Call-ID: <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>CSeq: 102 INVITE</p>
<p>User-Agent: X-Lite release 1104o stamp 56125</p>
<p>Content-Length: 0</p>
<p> </p>
<p><-------------></p>
<p>--- (9 headers 0 lines) ---</p>
<p>-- SIP/adf-00794e30 is ringing</p>
<p>ast-server*CLI> </p>
<p><--- SIP read from <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK5acbd17f;rport=5678</p>
<p>Contact: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588></p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350</p>
<p>From: "pepsi coke"<<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>>;tag=as12245807</p>
<p>Call-ID: <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>CSeq: 102 INVITE</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</p>
<p>Content-Type: application/sdp</p>
<p>User-Agent: X-Lite release 1104o stamp 56125</p>
<p>Content-Length: 185</p>
<p>v=0</p>
<p>o=- 2 2 IN IP4 192.168.0.12</p>
<p>s=CounterPath X-Lite 3.0</p>
<p>c=IN IP4 192.168.0.12</p>
<p>t=0 0</p>
<p>m=audio 15956 RTP/AVP 8 0 101</p>
<p>a=fmtp:101 0-15</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=sendrecv</p>
<p><-------------></p>
<p>--- (11 headers 9 lines) ---</p>
<p>Found RTP audio format 8</p>
<p>Found RTP audio format 0</p>
<p>Found RTP audio format 101</p>
<p>Peer audio RTP is at port <a href="http://192.168.0.12:15956">192.168.0.12:15956</a></p>
<p>Found description format telephone-event for ID 101</p>
<p>Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)</p>
<p>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</p>
<p>Peer audio RTP is at port <a href="http://192.168.0.12:15956">192.168.0.12:15956</a></p>
<p>list_route: hop: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588></p>
<p>[Jul 21 11:06:38] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>set_destination: Parsing <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588> for address/port to send to</p>
<p>set_destination: set destination to 116.18.35.235, port 28614</p>
<p>Transmitting (NAT) to <a href="http://116.18.35.235:28614">116.18.35.235:28614</a>:</p>
<p>ACK sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK00fdcc7c;rport</p>
<p>From: "pepsi coke" <<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>>;tag=as12245807</p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350</p>
<p>Contact: <<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>></p>
<p>Call-ID: <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>CSeq: 102 ACK</p>
<p>User-Agent: Asterisk PBX</p>
<p>Max-Forwards: 70</p>
<p>Content-Length: 0</p>
<p> </p>
<p>---</p>
<p>-- Call on SIP/adf-00794e30 left from hold</p>
<p>-- SIP/adf-00794e30 answered SIP/pepsi-9fd06cc0</p>
<p>ast-server*CLI> </p>
<p><--- SIP read from <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p> </p>
<p> </p>
<p><-------------></p>
<p>--- (0 headers 1 lines) ---</p>
<p>ast-server*CLI> </p>
<p><--- SIP read from <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p>SUBSCRIBE <a href="http://sip:adf@ast-server.axvoice.com:5678">sip:adf@ast-server.axvoice.com:5678</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 192.168.0.12:28614;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;rport</p>
<p>Max-Forwards: 70</p>
<p>Contact: <<a href="http://sip:adf@116.18.35.235:28614">sip:adf@116.18.35.235:28614</a>></p>
<p>To: "adf"<<a href="http://sip:adf@ast-server.axvoice.com:5678">sip:adf@ast-server.axvoice.com:5678</a>></p>
<p>From: "adf"<<a href="http://sip:adf@ast-server.axvoice.com:5678">sip:adf@ast-server.axvoice.com:5678</a>>;tag=5d297f22</p>
<p>Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.</p>
<p>CSeq: 1 SUBSCRIBE</p>
<p>Expires: 300</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</p>
<p>User-Agent: X-Lite release 1104o stamp 56125</p>
<p>Event: message-summary</p>
<p>Content-Length: 0</p>
<p> </p>
<p><-------------></p>
<p>--- (13 headers 0 lines) ---</p>
<p>Creating new subscription</p>
<p>Sending to 116.18.35.235 : 28614 (NAT)</p>
<p>Found peer 'adf'</p>
<p>Looking for adf in uscan_int (domain <a href="http://ast-server.axvoice.com">ast-server.axvoice.com</a>)</p>
<p>ast-server*CLI> </p>
<p><--- Transmitting (NAT) to <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p>SIP/2.0 404 Not Found</p>
<p>Via: SIP/2.0/UDP 192.168.0.12:28614;branch=z9hG4bK-d8754z-7039d4338568107f-1---d8754z-;received=116.18.35.235;rport=28614</p>
<p>From: "adf"<<a href="http://sip:adf@ast-server.axvoice.com:5678">sip:adf@ast-server.axvoice.com:5678</a>>;tag=5d297f22</p>
<p>To: "adf"<<a href="http://sip:adf@ast-server.axvoice.com:5678">sip:adf@ast-server.axvoice.com:5678</a>>;tag=as724c598c</p>
<p>Call-ID: MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.</p>
<p>CSeq: 1 SUBSCRIBE</p>
<p>User-Agent: Asterisk PBX</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</p>
<p>Supported: replaces</p>
<p>Content-Length: 0</p>
<p> </p>
<p><------------></p>
<p>Really destroying SIP dialog 'MTE5N2M4ZDY1OWRjOGQwMjgyOWEzZjkzYjA3Y2RkYWY.' Method: SUBSCRIBE</p>
<p>Reliably Transmitting (NAT) to <a href="http://116.18.35.235:28614">116.18.35.235:28614</a>:</p>
<p>OPTIONS sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport</p>
<p>From: "asterisk" <<a href="http://sip:asterisk@79.80.154.99:5678">sip:asterisk@79.80.154.99:5678</a>>;tag=as223ef4a7</p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588></p>
<p>Contact: <<a href="http://sip:asterisk@79.80.154.99:5678">sip:asterisk@79.80.154.99:5678</a>></p>
<p>Call-ID: <a href="mailto:5c66fbdf4234deca50d5c44a18641582@79.80.154.99">5c66fbdf4234deca50d5c44a18641582@79.80.154.99</a></p>
<p>CSeq: 102 OPTIONS</p>
<p>User-Agent: Asterisk PBX</p>
<p>Max-Forwards: 70</p>
<p>Date: Wed, 21 Jul 2010 15:07:07 GMT</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</p>
<p>Supported: replaces</p>
<p>Content-Length: 0</p>
<p> </p>
<p>---</p>
<p>ast-server*CLI> </p>
<p><--- SIP read from <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK42fde971;rport=5678</p>
<p>Contact: <sip:<a href="http://192.168.0.12:28614">192.168.0.12:28614</a>></p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=15133f38</p>
<p>From: "asterisk"<<a href="http://sip:asterisk@79.80.154.99:5678">sip:asterisk@79.80.154.99:5678</a>>;tag=as223ef4a7</p>
<p>Call-ID: <a href="mailto:5c66fbdf4234deca50d5c44a18641582@79.80.154.99">5c66fbdf4234deca50d5c44a18641582@79.80.154.99</a></p>
<p>CSeq: 102 OPTIONS</p>
<p>Accept: application/sdp</p>
<p>Accept-Language: en</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</p>
<p>User-Agent: X-Lite release 1104o stamp 56125</p>
<p>Content-Length: 0</p>
<p> </p>
<p><-------------></p>
<p>--- (12 headers 0 lines) ---</p>
<p>Really destroying SIP dialog '<a href="mailto:5c66fbdf4234deca50d5c44a18641582@79.80.154.99">5c66fbdf4234deca50d5c44a18641582@79.80.154.99</a>' Method: OPTIONS</p>
<p>ast-server*CLI> </p>
<p><------------></p>
<p>Scheduling destruction of SIP dialog '<a href="mailto:6514fece69f1718e5cefe72632909c0e@79.80.154.99">6514fece69f1718e5cefe72632909c0e@79.80.154.99</a>' in 23936 ms (Method: NOTIFY)</p>
<p>Reliably Transmitting (NAT) to <a href="http://116.18.35.235:28614">116.18.35.235:28614</a>:</p>
<p>NOTIFY sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport</p>
<p>From: "asterisk" <<a href="http://sip:asterisk@79.80.154.99:5678">sip:asterisk@79.80.154.99:5678</a>>;tag=as756cae64</p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588></p>
<p>Contact: <<a href="http://sip:asterisk@79.80.154.99:5678">sip:asterisk@79.80.154.99:5678</a>></p>
<p>Call-ID: <a href="mailto:6514fece69f1718e5cefe72632909c0e@79.80.154.99">6514fece69f1718e5cefe72632909c0e@79.80.154.99</a></p>
<p>CSeq: 102 NOTIFY</p>
<p>User-Agent: Asterisk PBX</p>
<p>Max-Forwards: 70</p>
<p>Event: message-summary</p>
<p>Content-Type: application/simple-message-summary</p>
<p>Content-Length: 92</p>
<p>Messages-Waiting: no</p>
<p>Message-Account: <a href="mailto:sip%3Aasterisk@79.80.154.99">sip:asterisk@79.80.154.99</a></p>
<p>Voice-Message: 0/0 (0/0)</p>
<p>---</p>
<p>ast-server*CLI> </p>
<p><--- SIP read from <a href="http://116.18.35.235:28614">116.18.35.235:28614</a> ---></p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK218cf73a;rport=5678</p>
<p>Contact: <sip:<a href="http://192.168.0.12:28614">192.168.0.12:28614</a>></p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=b9541904</p>
<p>From: "asterisk"<<a href="http://sip:asterisk@79.80.154.99:5678">sip:asterisk@79.80.154.99:5678</a>>;tag=as756cae64</p>
<p>Call-ID: <a href="mailto:6514fece69f1718e5cefe72632909c0e@79.80.154.99">6514fece69f1718e5cefe72632909c0e@79.80.154.99</a></p>
<p>CSeq: 102 NOTIFY</p>
<p>User-Agent: X-Lite release 1104o stamp 56125</p>
<p>Content-Length: 0</p>
<p> </p>
<p><-------------></p>
<p>--- (9 headers 0 lines) ---</p>
<p>Really destroying SIP dialog '<a href="mailto:6514fece69f1718e5cefe72632909c0e@79.80.154.99">6514fece69f1718e5cefe72632909c0e@79.80.154.99</a>' Method: NOTIFY</p>
<p>[Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:3074 update_call_counter: Call to peer 'adf' removed from call limit 2</p>
<p>Scheduling destruction of SIP dialog '<a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a>' in 18624 ms (Method: INVITE)</p>
<p>[Jul 21 11:07:15] DEBUG[23749]: chan_sip.c:5695 reqprep: Strict routing enforced for session <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>set_destination: Parsing <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588> for address/port to send to</p>
<p>set_destination: set destination to 116.18.35.235, port 28614</p>
<p>Reliably Transmitting (NAT) to <a href="http://116.18.35.235:28614">116.18.35.235:28614</a>:</p>
<p>BYE sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588 SIP/2.0</p>
<p>Via: SIP/2.0/UDP 79.80.154.99:5678;branch=z9hG4bK05cc42e6;rport</p>
<p>From: "pepsi coke" <<a href="http://sip:12345678901@79.80.154.99:5678">sip:12345678901@79.80.154.99:5678</a>>;tag=as12245807</p>
<p>To: <sip:adf@116.18.35.235:28614;rinstance=0266b8b94f488588>;tag=bd6f2350</p>
<p>Call-ID: <a href="mailto:25a6e3604896da0e5482a7565560ce3b@79.80.154.99">25a6e3604896da0e5482a7565560ce3b@79.80.154.99</a></p>
<p>CSeq: 103 BYE</p>
<p>User-Agent: Asterisk PBX</p>
<p>Max-Forwards: 70</p>
<p>Content-Length: 0</p></span> </div>
<div> </div>
<div>>Date: Wed, 28 Jul 2010 09:36:51 -0700<br>>From: Jim Dickenson <<a href="mailto:dickenson@cfmc.com">dickenson@cfmc.com</a>><br>>Subject: Re: [asterisk-users] Nat issue one way audio on IP dial<br>>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>>Message-ID: <<a href="mailto:353D789C-0987-49E1-988E-F3C98E41F8A7@cfmc.com">353D789C-0987-49E1-988E-F3C98E41F8A7@cfmc.com</a>><br>
>Content-Type: text/plain; charset="us-ascii"</div>
<p>>Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP >debug output a 192.168 address for the RTP packets to go to which of course will not work.<br>
>--<br>>Jim Dickenson<br>><a href="mailto:dickenson@cfmc.com">mailto:dickenson@cfmc.com</a></p>
<p>>CfMC<br>><a href="http://www.cfmc.com/">http://www.cfmc.com/</a></p>