Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.<br>
<br><br><div><span class="gmail_quote">On 21/07/06, <b class="gmail_sendername">Marco Mouta</b> <<a href="mailto:marco.mouta@gmail.com">marco.mouta@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Did you port forwar in your router RTP ports ? 10000-20000 to your *Box ?<br><br>On 7/21/06, Jose Limeres <<a href="mailto:jlimeres@gmail.com">jlimeres@gmail.com</a>> wrote:<br>> Hi,<br>><br>> I am experiencing a hard to solve problem with my VoIP provider. I can make
<br>> calls without any problem but I can not receive any. Actually, calls arive<br>> to * but the phone just does not ring. I believe must be a problem with NAT<br>> but I think I have a good config:<br>> - Extensions have nat=always and qualify=yes
<br>> - Have introduced in sip.conf Externip and localnet<br>> - ADSL modem/router is redirected to my * server<br>> - With sip debug I can see the call arrives<br>> Am I misssing something that someone else can see?
<br>><br>> Appreciate any hint. Thanks<br>> ==============================<br>> ======<br>> ASTERISK VERSION:<br>> Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q<br>><br>> SIP DEBUG CAPTURE<br>> <-- SIP read from
<a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> INVITE sip:34700758288001@87.218.175.120:5060 SIP/2.0<br>> Record-Route: <sip:<br>> <a href="http://62.22.20.194">62.22.20.194</a>;ftag=08ff6000ff05ff10ff00000e0c4effff;lr>
<br>> Via: SIP/2.0/UDP<br>> <a href="http://62.22.20.194">62.22.20.194</a>;branch=z9hG4bK90bf.b9c560e1.0<br>> Via: SIP/2.0/UDP<br>> <a href="http://62.22.20.207:5060">62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff
<br>><br>> From:<br>> <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff<br>> To: <<br>> sip:34700758288001@62.22.20.194:5060;user=phone>
<br>> Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> CSeq: 1 INVITE<br>> Contact: <<br>> <a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone><br>> Max-Forwards: 9
<br>> User-Agent: MERA MSIP v.1.0.2<br>> Cisco-Guid: 908093991-393679323-3151091529-1429652222<br>> Content-Type: application/sdp<br>> Content-Length: 216<br>><br>><br>> v=0<br>> o=- 1153435071 1153435071 IN IP4
<a href="http://62.22.20.207">62.22.20.207</a><br>> s=-<br>> c=IN IP4<br>> <a href="http://62.22.20.207">62.22.20.207</a><br>> t=0 0<br>> m=audio 59320 RTP/AVP 18 4 101<br>> a=rtpmap:18 G729/8000<br>> a=rtpmap:4 G723/8000
<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-15<br>><br>> --- (14 headers 10 lines)---<br>> Using INVITE request as basis request -<br>> d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> Sending to
<a href="http://62.22.20.194">62.22.20.194</a> : 5060 (non-NAT)<br>> Found peer 'Peoplecall'<br>><br>> Reliably Transmitting (NAT) to <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> SIP/2.0 407 Proxy Authentication Required
<br>> Via: SIP/2.0/UDP<br>> <a href="http://62.22.20.194">62.22.20.194</a>;branch=z9hG4bK90bf.b9c560e1.0;received=<br>> <a href="http://62.22.20.194">62.22.20.194</a><br>> Via: SIP/2.0/UDP<br>> <a href="http://62.22.20.207:5060">
62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff<br>> From: <<br>> <a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
<br>> To: <<a href="mailto:sip:34700758288001@62.22.20.194">sip:34700758288001@62.22.20.194</a><br>> :5060;user=phone>;tag=as476d14de<br>> Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> CSeq: 1 INVITE
<br>> User-Agent: Asterisk PBX<br>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> Contact: <<br>> <a href="mailto:sip:34700758288001@87.218.175.74">sip:34700758288001@87.218.175.74
</a>><br>> Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0"<br>><br>> Content-Length: 0<br>><br>><br>> ---<br>> Scheduling destruction of call<br>> 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1
' in 15000 ms<br>> asterisk1*CLI><br>> <-- SIP read from<br>> <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> ACK sip:34700758288001@87.218.175.120:5060 SIP/2.0<br>> Via: SIP/2.0/UDP <a href="http://62.22.20.194">
62.22.20.194</a>;branch=<br>> z9hG4bK90bf.b9c560e1.0<br>> From:<br>> <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff<br>><br>> Call-ID:
d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> To:<br>> <sip:34700758288001@62.22.20.194:5060;user=phone>;tag=as476d14de<br>> CSeq: 1 ACK<br>> User-Agent: OpenSer (1.0.0 (i386/linux))<br>> Content-Length: 0
<br>><br>><br>><br>> --- (8 headers 0 lines)---<br>> REGISTER 13 headers, 0 lines<br>> Reliably Transmitting (no NAT) to <a href="http://62.22.20.194:5060">62.22.20.194:5060</a><br>> :<br>> REGISTER sip:
<a href="http://sip.peoplecall.com">sip.peoplecall.com</a> SIP/2.0<br>> Via: SIP/2.0/UDP<br>> <a href="http://87.218.175.74:5060">87.218.175.74:5060</a>;branch=z9hG4bK4a6abe4f;rport<br>> From: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">
sip:34700758288001@sip.peoplecall.com</a><br>> >;tag=as79fdfc26<br>> To: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>><br>> Call-ID:<br>> <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">
1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a><br>> CSeq: 421 REGISTER<br>> User-Agent: Asterisk PBX<br>> Max-Forwards: 70<br>> Authorization: Digest username="34700758288001", realm="<br>> <a href="http://sip.peoplecall.com">
sip.peoplecall.com</a>", algorithm=MD5, uri="sip:<a href="http://sip.peoplecall.com">sip.peoplecall.com</a><br>> ", nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6",<br>> response="ee782a37bae7eed1a0a881147c733ede", opaque=""
<br>><br>> Expires: 120<br>> Contact: <<a href="mailto:sip:34700758288001@87.218.175.74">sip:34700758288001@87.218.175.74</a>><br>> Event: registration<br>><br>> Content-Length: 0<br>><br>><br>
> ---<br>> asterisk1*CLI><br>> <-- SIP read from <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> SIP/2.0 200 OK<br>><br>> Via: SIP/2.0/UDP<br>> <a href="http://192.168.1.104:5060">
192.168.1.104:5060</a>;branch=z9hG4bK4a6abe4f;rport=5060<br>> From: <<br>> <a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>>;tag=as79fdfc26<br>> To: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">
sip:34700758288001@sip.peoplecall.com</a><br>> >;tag=555271b30cfd40f8a3b4837b054360a3.975d<br>> Call-ID: <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a><br>
><br>> CSeq: 421 REGISTER<br>> Contact:<br>> <sip:34700758288001@192.168.1.104:5060>;expires=120<br>> Server: OpenSer (1.0.0 (i386/linux))<br>> Content-Length: 0<br>><br>><br>> --- (9 headers 0 lines)---
<br>> Scheduling destruction of call '<br>> <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a>' in 32000 ms<br>> Destroying call 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1
<br>> '<br>> asterisk1*CLI> sip no debug<br>> SIP Debugging Disabled<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">
Easynews.com</a> --<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>><br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br>><br>><br>><br><br><br>--<br>Com os melhores cumprimentos,<br><br>Marco Mouta<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com
</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>
</blockquote></div><br>