On Tue, 2008-02-26 at 12:31 -0500, Dirk Enrique Seiffert wrote: > I aquired an account with a reseller net-voz.com: I did some testing with > the account directly from a Snom300 phone - works without a problem, > (behind the nat) I spent hours testing and adjusting the trunk > configuration for net-voz, maybe sombody on the list can give a helpful hint:
I'll take a stab at it. > First of all: Registry works! Registering to another host doesn't mean anything when it comes to sending them a call. Registration only tells them your IP address and port so that they can send calls *to you*. > On the astersik CLI the logs show: > > Audio is at 192.168.8.3 port 14800 > Adding codec 0x4 (ulaw) to SDP > Adding codec 0x8 (alaw) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 190.144.151.212:5060: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport > From: "901" <sip:[EMAIL PROTECTED]>;tag=as3c6dfee5 > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", > algorithm=MD5, uri="sip:[EMAIL PROTECTED]", > nonce="120404195526111105702055508208", > response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque="" > Date: Tue, 26 Feb 2008 16:09:09 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 260 > > v=0 > o=root 2381 2382 IN IP4 192.168.8.3 > s=session > c=IN IP4 192.168.8.3 > t=0 0 > m=audio 14800 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- Notice how the Contact Header and the SDP all have the IP address of 192.168.8.3? If your firewall isn't masquerading (rewriting) those addresses as the SIP traffic goes through it, then the device on the other end is going to try to contact 192.168.8.3, and I'm guessing it's going to have a hard time doing that. (This would also explain why you're seeing outbound traffic only in your tcpdump traces.) -- Jared Smith Community Relations Manager Digium, Inc. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
