[Jaco]

Dear friends,

Dear sirs,

i am a SPA 3000 user from a long time and inhave been using most of the features of this wonderful little machine without any problem.

now i have received the LINSKYS SPA3102 and i have lot of problems with the PSTN - VOIP FEATURE.

whenever i call the PSTN and i get the VOIP dial tone, after dialing i dont get the call to be done a long silence period remains without the call to be made.

if anyone already configured the SPA3102 with DELTATHREE DID accounts please be so kind in sharing send me the correct configuration parameters in order to make this device work properly.

THANKING IN ADVANCE

Jacobo

[tommycr]

I need your help ! I use to have the Linksys PAP-2 with ICH account for several time. Now same as you, I bought the SIP 3102. Now I am in troubles: I can receive calls with no problems , but I am still unable to make calls thru Line1. Could you give some "TIPS" on how to configure the Line1 tab ?
PSTN will be the next project .. je je


Please take note that, My NAT is disable.

I will appreciate your help ! Could you give me some guides according your actual SIP 3102 configuration Page ?

[hwittenb]

Originally Posted by tommycr

I can receive calls with no problems , but I am still unable to make calls thru Line1.

If you can receive calls from ICH you are close to solution. Why don't you save your configuration on your hard drive, make a .zip file of it, and post it here for comment.

Please take note that, My NAT is disable.

Your 3102 is attached directly to the modem or does this mean something else?

[tommycr]

My SIP 3102 unit is attached to the modem with a Linksys WRT54G
with this router I already opened ports 5060-5063.

Please find attached my configuration FILE.

I am not configuring the PSTN tab yet until I solve the "outgoing calls issue"

THANKS FOR YOUR HELP

[hwittenb]

I reviewed your configuration and these are my thoughts.

Line 1 shows registered. That's good.

On the SIP tab you have STUN enable YES but no STUN server. For the server, I would put stun.softjoys.com. You can use any STUN server you wish. It echos back STUN commands showing the port and ip address from which the message was received.

On the Line 1 tab NAT Keep Alive Message should be $NOTIFY.
On the Line 1 tab NAT Keep Alive Dest should be $PROXY.

You are using an outbound proxy. I would try it first without using an outbound proxy by clearing the outbound proxy field and setting Use Outbound Proxy No. If this does not work and you need to use an outbound proxy, then on the SIP tab I would set STUN enable to NO, although it may work OK enabled.

You are using an AUTH ID. Usually this is not necessary. I would try it both with and without those field filled in. Usually your userid is sufficient. If the field is not needed, I wouldn't use it.

You are not using the PSTN line at present. In the future if you do want to use it you will need to fill in something in the voip user.

If you use a STUN server you generally do not need to forward any ports on your router. I would start by trying to use a STUN server without any ports forwarded. If you have to use an outbound proxy, you may need to forward some ports. You may need to forward your incoming SIP signalling ports plus you should forward the ports used by the voice packets which are the RTP ports which you currently have set from 16384 to 16482.

[tommycr]

THANKS A LOT FOR YOUR COMMENTS ON MY CONFIGURATION FILE:
LOOK AT MY COMMENTS IN CAPS BELOW...

Originally Posted by hwittenb

I reviewed your configuration and these are my thoughts.

Line 1 shows registered. That's good.

On the SIP tab you have STUN enable YES but no STUN server. For the server, I would put stun.softjoys.com. You can use any STUN server you wish. It echos back STUN commands showing the port and ip address from which the message was received.


I MADE THIS CHANGES, WITH AND WITHOUT STUN OPTIONS ONE BY ONE, BUT DOESN´T WORKS

On the Line 1 tab NAT Keep Alive Message should be $NOTIFY.
On the Line 1 tab NAT Keep Alive Dest should be $PROXY.

You are using an outbound proxy. I would try it first without using an outbound proxy by clearing the outbound proxy field and setting Use Outbound Proxy No. If this does not work and you need to use an outbound proxy, then on the SIP tab I would set STUN enable to NO, although it may work OK enabled.

I ACTIVATED AN DEACTIVATED THE OUTBOUND PROXY FIELDS WITHOUT SUCESS.

AFTER THAT, I ACTIVATE AGAIN AND DEACTIVATE THE STUN SERVER BUT DOESN´T LIKE TO WORK NEITHER.......


You are using an AUTH ID. Usually this is not necessary. I would try it both with and without those field filled in. Usually your userid is sufficient. If the field is not needed, I wouldn't use it.

WHEN I CLEAN THIS FIELDS, I ALLWAYS RECEIVE THE MESSAGE: "YOUR ACCOUNT IS TEMPORARILY UNAVAILABLE. PLEASE TRY AGAIN LATER. IF THE PROBLEM PERSISTS, PLEASE CONTACT CUSTOMER SERVICE. 401"

I ALSO TESTED WITH WRONG NUMBERS ON THE AUTH ID FIELD, AND I ALLWAYS RECEIVED THE SAME MESSAGE ..... CONCLUSION: WHEN AUTH ID FIELD IS BLANK OR WRONG ......ERROR MESSAGE. WHEN "USE AUTH ID FIELD" IS SET TO "NO" ....ERROR MESSAGE TOO.


You are not using the PSTN line at present. In the future if you do want to use it you will need to fill in something in the voip user.

WHEN I HAVE DIAL TONE , I WILL SETUP THE PSTN ...............


If you use a STUN server you generally do not need to forward any ports on your router. I would start by trying to use a STUN server without any ports forwarded. If you have to use an outbound proxy, you may need to forward some ports. You may need to forward your incoming SIP signalling ports plus you should forward the ports used by the voice packets which are the RTP ports which you currently have set from 16384 to 16482.

IN ORDER TO BE SURE THE CORRECT PORTS ARE OPEN ON MY ROUTER, I HAVE TESTED PUTTING MY SIP-3102 UNDER THE DMZ (DESMILITARIZED ZONE) WITH NO SUCCESS NEITHER..... I RETURNED BACK IN ORDER TO PREVENT UNATHORIZED ACCESS TO THE SIP 3102 THROUGH MY ROUTER....

SEEMS TO BE SOMETHING VERY SIMPLE...... ( SOMETHING LIKE "ALLWAYS PRESS AN STAR BEFORE DIAL .... )

THANKS A LOT !

[hwittenb]

The next step is to run the Sipura debug server and report the results. You need to download a program from Sipura, and make some 3102 configuration changes to output the debug information which will be stored on a file on your pc. Startup the program, making sure you aren't running any firewall that stops the pc from receiving the information. After running some test calls, stop the program, make a zip file of the results and attach it to a posting.

You dowload the Sipura program here: SPA adaptors Frequently Asked Questions (FAQ)
You need to determine the internal network address of your pc. You can do this by running ipconfig. Put the pc's ip address under Debug Server on the 3102 System tab. Set the Debug Level to 3. On the Line 1 tab, set the SIP Debug Option to Full. When you startup the downloaded Syslog program, and you take the phone off hook you should see data go by if it is working properly. This data is stored on a file on your hard drive.

[dibsmft]

You should also open/forward some RTP ports. 8000 - 8020 should be enough.

[tommycr]

you mean change the actual 16384 to 16482 for 8000 to 8100. parameters on the SIP tab ? or open this range in the router ? Or both ?

I already check with the first option and the second option is no applicable since I already had the DMZ open for my SIP.....

[dibsmft]

You need a range of ports open for rtp ... you did not indicate that any were open. I do not normally run STUN.

Try this....

Get an account with voxalot.org

Put ICH in as your provider and see if calls work OK then. This seems to work with most provider on my setup(s). Another method might be to use the 3102 as router (it seems to work very well) and use the Linksys WRT54G as a bridge/hub. Then the 3102 will have your open network address and you should have fewer problems with it. It seems that you are not the only one that has problems with the Linksys WRT54G in this area.
Try Voxalot first as you do not need to make any network changes for that.