[asterisk-users] Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely

[LATEEF, IRFAN (ATTSI)]

Hi,
I am try to configure Asterisk as PBX system with two interfaces as
shown below. One interface pointing to the local subnet with a SIP phone
and another interface pointing to the external ISP SIP Sever. 
SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external-
intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld

I am able to setup a call from the Phone to the outside world and I have
the audio (RTP packets) coming from the outside world being routed to my
phone 
but the audio from my Phone IP(X.X) is not going out to the SIP-Server.
In fact I think it is not even reaching the Asterisk server because the
SDP in the 183 going to the phone has the IP address of the
external-inf(Z.Z.247.106) of the Asterisk PBX when it should actually
(Y.Y.47.149)

		<--- Transmitting (NAT) to X.X.141.32:5060 --->
		SIP/2.0 183 Session Progress^M
		Via: SIP/2.0/UDP
X.X.141.32;branch=z9hG4bK87468d20000002f44b86a00400006f2b00000166;receiv
ed=X.X.141.32;rport=5060^M
		From: "Irfan Lateef"
<sip:2005 at Y.Y.47.149>;tag=327f290e2e7^M
		To: <sip:99084611234 at Y.Y.47.149>;tag=as24228e21^M
		Call-ID: 876BAA6B36F644F7B4EF7BE5D4B7E8BD0x87468d20^M
		CSeq: 2 INVITE^M
		User-Agent: Asterisk PBX 1.6.0.17^M
		Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY, INFO^M
		Supported: replaces, timer^M
		Require: timer^M
		Session-Expires: -1;refresher=uas^M
		Contact: <sip:99084611234 at Z.Z.247.106>^M
		Content-Type: application/sdp^M
		Content-Length: 315^M
		^M
		v=0^M
		o=root 1021147583 1021147583 IN IP4 Z.Z.247.106^M
		s=Asterisk PBX 1.6.0.17^M
		c=IN IP4 Z.Z.247.106^M
		t=0 0^M
		m=audio 18702 RTP/AVP 0 8 3 101^M


I have the following in the sip_nat.conf

		localnet=Y.Y.47.149/255.255.0.0
		externhost=Z.Z.247.106
		externrefresh=10
		fromdomain=att.com
		nat=yes
		qualify=yes
		canreinvite=no


I think the SDP should have give the Y.Y.47.149 IP on the local net side
to the phone. But I am unable to figure how make it do that.

The Asterisk log shows this.
		 [Feb 25 11:06:30] VERBOSE[1449] logger.c:     --
Executing [s at macro-dialout-trunk:19]
^[[1;36;40mDial^[[0;37;40m("^[[1;35;40mSIP/2005-19dc0db8^[[0;37;40m",
"^[[1;35;40mSIP/ATT-alpi016-IPFlex1/19084611234,300,^[[0;37;40m") in new
stack
		[Feb 25 11:06:30] VERBOSE[1449] logger.c:   == Using SIP
RTP TOS bits 184
		[Feb 25 11:06:30] VERBOSE[1449] logger.c:   == Using SIP
RTP CoS mark 5
		[Feb 25 11:06:30] VERBOSE[1449] logger.c:     -- Called
ATT-alpi016-IPFlex1/19084611234
		[Feb 25 11:06:32] VERBOSE[1449] logger.c:     --
SIP/ATT-alpi016-IPFlex1-19dda0f8 is making progress passing it to
SIP/2005-19dc0db8
		[Feb 25 11:06:32] VERBOSE[1449] logger.c: Audio is at
Z.Z.247.106 port 18702
		[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec
0x4 (ulaw) to SDP
		[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec
0x8 (alaw) to SDP
		[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding codec
0x2 (gsm) to SDP
		[Feb 25 11:06:32] VERBOSE[1449] logger.c: Adding
non-codec 0x1 (telephone-event) to SDP
		[Feb 25 11:06:32] VERBOSE[1449] logger.c:


Any help is greatly appreciated.

Thanks and Regards,
Irfan Lateef