[anischaudry]

Hello Gurus,

I need a very simple hot line setup between two points A and B, both will have SPA3K and are connected to Internet with ADSL.

Location A and B have dynamic Public IPs on the WAN and Static IP assigned to both the SPAs. At one end the ISP has blocked the ports used for VOIP (5060-5061). Also unable to register with FWD or any other, this was working perfectly fine few months back.

Would like to dial from A location for location B to ring, and dial from Location B for location A to ring.

I have reset both the SPAs to factory default what would be the configuration at each location. Is it possible to use any other UDP port than SIP Port

A detailed response would be very much appreciated .

Thank you in advance.

Anis

[hwittenb]

It is possible to use any port numbers you wish. You just need to keep track of what they are and put the port number in the address that you are calling. This is the port number that is used for the sip signalling. Another set of port numbers is used for the rtp audio packet stream. The port numbers used here are setup on the sip tab. You can also use any range you wish here too. Each side notifies the other of the rtp port to use during the call setup.

A hot line would dial the distant ata when the attached analog telephone goes off hook. On the distant ata you could pickup the phone and talk to the caller. For a simple hot line between two points you would setup Line 1:
Line Enable: YES
Register: NO
Make Call Without Reg: YES
Ans Call Without Reg: YES
Codec: Settings should be the same for the two units.

Dial Plan: (<:xxx.xxx.xxx.xxx: pppp>S0) where xxx is the ip address, pppp is the port number of the ata that you are calling. S0 (S zero) says send the dial string immediately.
(note: this forum forces you to put a space character after the colon ( : ). Do not put a space in the actual address.)

If for some reason you have an entry in the userid field you need to include that in the dial plan address: (<:userid@xxx.xxx.xxx.xxx: pppp>S0)

The next problem you need to overcome is the transversal of your local networks. There is no set solution for this. It will depend on the internet sharing router that you are using. The problem to overcome is that you are sending the packets to the distant external ip address and the port number of your ata. The packets must go thru the distant router and then reach the ata using the indicated port number. This is a problem for both the sip signalling and for the rtp packets. It is the source of the common "I can't hear" complaint.

You need to forward the sip signalling port number in each router to the locally attached ata. This is because the ata does not register and the incoming call is not solicited. The router needs to know where to send the incoming sip packet when it is received. You also may need to forward the rtp port range.

First
Try it like it is

Second try
Set NAT Mapping Enable: YES
Set NAT Keep Alive Enable: YES (optional. Probably don't need this)

Third try
Try using an external STUN server. NAT Mapping Enable YES must be set to use STUN.
On the Sip Tab
Substitute VIA Addr:YES
Send Resp To Src Port:YES
STUN Enable:YES
STUN Server:stun.fwdnet.net (or any other STUN server)

Fourth try
Port forward the rtp ports in your router to the ata local address if you haven't done it already.

Fifth try
Put the ata in the router's DMZ

[anischaudry]

First of all thanks for taking time to provide a solution to my problem. I really appreciate your help.

I have tried all the 5 steps you have mentioned. I first tried with UDP port 257 at both end for line #1. and special care was taken to put a space before the port and : in the dial plan. No rings at either end, although the web page and syslog says it is dialling.

Later I changed the UDP port at both ends to 27910. At both end it is hissing sound when the phone is lifted and on either end it does not ring. The ports are open on both the router. Both ATA's IP is as DMZ . Before selecting the Ports I tested with the "test port " utility and they are open at both ends.

Is it something to do the ports select or still missing some configuration.

[hwittenb]

Originally Posted by anischaudry

special care was taken to put a space before the port and : in the dial plan.

I didn't say it clearly, there should be no spaces between the address, the colon, and the port number.

I would favor the higher port numbers.

Is the test over the internet or behind the same router? If you are behind the same router I would change the rtp port range on one of the ata's so that it is different from the other one. The rtp port range is on the sip tab.

You can save your configurations to your hard drive, make a zip file of them and attach to a posting here.

[anischaudry]

The test is over the internet. Both the ATA are behind the router.

I can configure the remote system through VNC, which is working fine.


Please find attach the configuration of both the SP3K

[hwittenb]

I forgot to mention that you should save the configuration after logging in as admin/advanced. The remote config was OK, but I couldn't review the line 1 settings for the Auh SP3k ata.

I tried a call to the remote ata and it appeared to be successful. I ran a trace and it showed call completion.

Calling the Auh SP3k there was no response. Perhaps the spa was powered down. A ping was successful to the ip address which means the ping got to the router.

I noticed that port 27910 is the port used by people playing the "Quake II" multiuser computer game. I tried running a packet trace and my ethereal pc packet sniffer program thought I was running Quake 2. I had never seen that before, but I killed the protocol in the packet trace and things went back to normal.

Your configuration in the remote SPA3k looks OK. I would blank the entry in EXT SIP Port:, however it probably doesn't make any difference. I would also set Refer-To Target Contact: NO but again that probably doesn't make any difference. I would also get rid of the space between the ip address and the port number in the dial plan, but that also might not make any difference.

I would remove the cfwd no ans entry on the User 1 tab. Before this would give you a problem though the phone would have to ring for awhile.

If the configuration in the Auh Sp3k is similiar, then I would start running a trace on the systems to see if there is any response. Here are instructions for running the trace: http://forum.voxilla.com/linksys-sip...ogs-20736.html (SPA-3102 - system logs?)
be sure to set both debug level 3 and debug option full on the line 1 tab.

[anischaudry]

Sorry ! it was my mistake I should have posted the advance page for the SPA at my end.

Attached is the advance configuration and the syslog text file.

I really appreciate your valuable help in resolving this issue.

Thanks for your time and assistance.

[hwittenb]

The configuration looks OK. It should work.

[anischaudry]

unfortunately it does not, i have changed the ports to 27950 / 27920. It is the same hissing sound from both end.

Has it something to do the ports or ...

Thanks again

[Arsalan]

Anis, can I assume that one of your ATAs is located in a country which is blocking ports.

I am using 2 SPA3000 with Vbuzzer.com accounts configured on both ends for a similar purpose and use ports 5080 or 5090 on both sides. The setup works!!

Vbuzzer accounts are free if you want to call within the community.

I have also setup speedials on both the sides so that you can pick the line and dial #1 to connect to the other ATA. I also have one of the ATAs working as a PSTN gateway using another vbuzzer account.

You can configure vbuzzer on your ATA from Voxilla configuration page.