Hi Folks, I did not get any responses from the list, but I debugged and fixed it myself. Just in case anybody runs into this again here is the problem and the solution.
When there are two interfaces as in shown in my setup below. Most examples on the internet talk about peers behind firewalls and using NAT. This is a case where asterisk has a static IP on the external side and a multiple local subnets on the internal side with peers and asterisks on different local subnets. The key to the solution is to use the right mask in sip_nat.h . If this your problem read on ... The local interface IP as 130.Y.47.149 and the softphone was at 135.X.141.32 and the setting in the sip_nat.conf was as follows: localnet=130.X.47.0/255.255.255.0 externhost=Z.Z.247.106 When the SDP was being prepared for the local peer (softphone) the local peer's IP addr did not match local net address because the Mask was 255.255.255.0 and the first bytes were different. Hence asterisk thought that the peer was on the external side and therefore chose the external IP to send to the local peer in the SDP. In order to get the asterisk to recognize that 130 and 135 and in fact both on the local net the subnet mask had to chosen in such a way that only the bits common to both number were compared hence 130.0.0.0 mask and address resolves it correctly. Therefore I change the sip_nat.conf as below and it works. localnet=130.0.0.0/130.0.0.0 externhost=Z.Z.247.106 The following link was very helpful in understand the SIP NAT configuration and settings. http://www.linuxjournal.com/article/9399 This excerpt from the above link explains it. The simplest situation is when a SIP client is behind a NAT gateway connecting to a server on the Internet. The client creates the translation entry for the SIP traffic when it first registers. As long as there is frequent communication between the two hosts, such as one packet per minute, the channel will stay open. The only configuration needed is to have the client use its external address in all SDP packets. On clients that support it, enable STUN (Simple Traversal of UDP through NAT), so the client can determine the external address dynamically, or enter it manually. Asterisk doesn't support STUN at this time, so all NAT configuration must be done manually. The following commands in /etc/asterisk/sip.conf set up the NAT properly: [general] localnet=192.168.0.0/255.255.0.0 ; or your subnet externip=x.x.x.x ; use your address [YOURREMOTEPEER] ; your peer's name nat=yes qualify=yes ; Force keepalives And another article at the following link. http://www.voip-info.org/wiki/view/aster...@home+handbook+wiki+chapter+2 And this section from the above link really clarifies the use of localnet IP and mask usage. Here is our example: The DNS Name (or you can enter the IP Address) of the VOIP SIP provider is sip.voipcompany.com. The static LAN/Private IP address of your aster...@home server is 192.168.1.2. The static WAN/Public IP address of your router is 1.2.3.4 (if you use DynamicDNS, we'll use your domain name) Enter your router/firewall configuration and add: Forward UDP Port 5060-5082 to 192.168.1.2 Forward UDP Port 10000 to 20000 to 192.168.1.2 You also need to edit the sip_nat.conf file. Inside of AMP, click Maintenance ----> Config Edit ----> sip_nat.conf. Inside of sip_nat.conf add the following and click "Update": externip = 1.2.3.4 (enter your permanent WAN/Public internet address here. Or you if you have one, you can use a DynamicDNS domain name. Obviously it's easier to get a static IP address and avoid using DynamicDNS altogether.) If you are using a dynamic dns domain name, you should rather use: externhost = home.mydomain.com externrefresh = 5 (which means lookup hostname every 5 minutes to refresh ip adress) localnet = internal.network.address.0/255.255.255.0 (put your LAN/Private NETWORK address of your aster...@home server, this is NOT the IP address of the server!!!!) To determine your local NETWORK address (NOT the IP address!!) you have to know a little about your subnet mask (255.255.255.0 numbers). If the IP address of the aster...@home server is 192.168.1.5 255.255.255.0, then the NETWORK address is 192.168.1.0 If the IP address of the aster...@home server is 192.168.7.2 255.255.255.0, then the NETWORK address is 192.168.7.0 If the IP address of the aster...@home server is 192.168.100.84 255.255.255.0, then the NETWORK address is 192.168.100.0 Hope this helps !!! Regards, Irfan Lateef -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of LATEEF, IRFAN (ATTSI) Sent: Thursday, February 25, 2010 5:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.0.17 PBX with two interfaces does notroutes RTP packets - SIP Conf Problem likely Importance: High 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:2...@y.y.47.149>;tag=327f290e2e7^M To: <sip:99084611...@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:99084611...@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 [...@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 -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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