[asterisk-users] Sip trunk mystery

[Jared Smith]

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:5756646022 at sip.net-voz.com SIP/2.0
> Via: SIP/2.0/UDP 192.168.8.3:5060;branch=z9hG4bK0772982f;rport
> From: "901" <sip:5515816168 at sip.net-voz.com>;tag=as3c6dfee5
> To: <sip:5756646022 at sip.net-voz.com>
> Contact: <sip:5515816168 at 192.168.8.3>
> Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch",
> algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com",
> 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.