SPA 3102 Outgoing

[Piggy]

I have went through many helpful topics on the forum, but it did not resolve my problem. I am able to call extension-to-extension, receive PSTN-to-Voip calls properly. I am however unable to dial out after over a week of trying. I have followed the instructions from http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE. Here are information regarding my setup:

location: North America
SPA Firmware: 5.1.7(GW)
Hardware version: 1.1.5
SPA-3102: 192.168.2.2
3cx server: 192.168.2.101
PSTN virtual extension: 10008 (registered and online)
ATA extension: 888 (registered and online)
Error shown on 3cx:
19:51:09.441 CallConf::Rejected [CM103005] Call(2) is rejected: Not Found
19:51:09.441 StratInOut:nCancel [CM104008] Call(2): Call from Ext.101 to 16265551212 terminated; cause: 404 Not Found; from IP:192.168.2.2
19:51:09.332 CallConf:nIncoming [CM103002] Call(2): Incoming call from 101 (Ext.101) to sip:[email protected]
Configurations:
SPA Line1 http://www.krazykarz.net/images/spa-line1.pdf
SPA PSTN http://www.krazykarz.net/images/spa-pstn.pdf
3CX PSTNGateway http://www.krazykarz.net/images/3cx-pstngateway.pdf
3CX Extension 888 http://www.krazykarz.net/images/3cx-Ext888.pdf

I am suspecting that I setup Line1 wrong. According to the http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE instructions, on instruction item #18+ I should be setting up an Extension for ATA(for use on SPA Line1's subscriber info), which I assume will handle the outgoing voip-to-PSTN traffice (correct me if I'm wrong!) . I created Extension 888 (configuration shown on link above) on 3cx as a regular extension. Not sure if I did that part wrong. I get the error as show earlier. If I change the subscriber information on Line1 to the authention for extension 10008 (The virtual extension for PSTN gateway on 3cx), I would get a continuous ring... but it never dialed out of course. The 3cx log would then show:
22:31:21.118 StratInOut:nHangUp [CM104007] Call(23): Call from Ext.101 to 16265551212 has been terminated by Ext.101; cause: CANCEL; from IP:192.168.2.88
22:30:53.375 CallConf:nProvisional [CM103003] Call(23): Ln:10008@PSTN-Gate is ringing
22:30:53.282 CallConf:nIncoming [CM103002] Call(23): Incoming call from 101 (Ext.101) to sip:[email protected]
This simply led me to believe that it may have something to do with the subscriber info on Line1 of SPA and the extension for ATA (based on the configuation instructions) I am suppose to create on 3CX. Any ideas/help? Thanks everyone!

P.

 

[Patch]

Sigh! I've exactly the same problem of nanocomputers.

I tried several configurations but no success at all, I'm unable to call through the pstn line.
I can receive calls from pstn, Do calls to the phone connected to the SPA, receive calls from the phone connected to the SPA but no extensin can use PSTN Line.

I always get "404 NOT FOUND" error... The log is exactly the same as nanoncomputers' one.

It would be nice if someone could post a complete working configuration .

Thanks
PAtch :shock:

 

[jantoney]

Hey All!..

arrg i left my desk and didn't hit submit so i lost my previous post... so here goes a shorter version!.. :x

I had similar issues and mine ended up being a port issue...
Check under the 'GATEWAY/PROVIDER config of the linksys box IN 3CX that the port is set to 5060...

I didn't read the whole forum post so im not too sure if this has been said but the linksys config is a bit confusing and this took me a while to get sorted before it was working!
Hope it works for you...

Then in the linksys config go
Click ADMIN LOGIN
Click ADVANCED
Click VOICE (tab)
Click LINE 1 (tab)
set the SIP PORT to 5061 <-------- NOTICE 5061 NOT 5060
Click the VOICE (tab)
Click PSTN LINE (tab)
Set the SIP PORT to 5060 <--------NOTICE 5060 NOT 5061
Click SAVE ALL CHANGES

Reset the linksys by removeing the power and plugging it back in... this forces a re-register and ensures that the config was saved!

PLEASE LET THE BOARD KNOW HOW YOU GO and also post back any logs if it fails!!!

 

[nanocomputers]

hi,

still isn't working as the settings were allready like that in the linksys, though I pulled the power cord and reconnected it.

Could it be a problem in the outbound rules?

This is the log for the call:

19:11:15.687 Call::Terminate [CM503008]: Call(101): Call is terminated
19:11:15.609 Call::RouteFailed [CM503014]: Call(101): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
19:11:15.593 CallCtrl:nSelectRouteReq [CM503013]: Call(101): No known route to target: [sip:[email protected]]
19:11:08.125 CallCtrl:nIncomingCall [CM503001]: Call(101): Incoming call from Ext.1000 to [sip:[email protected]]

 

[jantoney]

Hey nanocomputers...
mmm looking at your logs compaired to a normal call to a good call log there are a few major things that stand out...

My settings
My extension = 400
Number to call = 0418603000
Prefix to get line out = 0
My Linksys PSTN box extension = 10002 and 10001 - I HAVE TWO OF THEM setup to use line 2 before line 1.

A good log
09:07:53.090 Call::Terminate :[CM503008]: Call(258): Call is terminated
09:07:53.090 Call::Terminate :[CM503008]: Call(258): Call is terminated
09:07:35.263 CallCtrl:nLegConnected :[CM503007]: Call(258): Device joined: sip:[email protected]:5060
09:07:35.263 CallCtrl:nLegConnected :[CM503007]: Call(258): Device joined: sip:[email protected]:5060
09:07:35.263 Line:rintEndpointInfo :[CM505002]: Gateway:[Line2] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.5(GWa)] Transport: [sip:192.168.1.19:5060]
09:07:35.169 CallCtrl:nSelectRouteReq :[CM503004]: Call(258): Calling: PSTNline:10001@[Dev:sip:[email protected]:5060]
09:07:35.153 CallCtrl:nSelectRouteReq :[CM503004]: Call(258): Calling: PSTNline:10002@[Dev:sip:[email protected]:5060]
09:07:35.122 CallCtrl:nIncomingCall :[CM503001]: Call(258): Incoming call from Ext.400 to [sip:[email protected]]

Looking at the logs
** Working from the bottom **
1 - call from EXT 400 to number 00418603000 (take prefix of 0 leaves the number to be dialed as 0418603000)
2 - calling the Linksys PSTN box
3 - Joined the devices (SIP:400 and SIP:10002)
4 - Call terminated because i hung up

Your Logs
19:11:15.687 Call::Terminate [CM503008]: Call(101): Call is terminated
19:11:15.609 Call::RouteFailed [CM503014]: Call(101): Attempt to reach [sip:[email protected]] failed. Reason: Not Found
19:11:15.593 CallCtrl:nSelectRouteReq [CM503013]: Call(101): No known route to target: [sip:[email protected]]
19:11:08.125 CallCtrl:nIncomingCall [CM503001]: Call(101): Incoming call from Ext.1000 to [sip:[email protected]]

Looking at your logs
** Working from the bottom **
1 - call from EXT 1000 to number 0614024168 (is your prefix 0??? because its not trying to call your PSTN box... its trying to call an extension with that EXT number.)
2 - the call is then stuffed from this point....

QUESTIONS
Can you please confirm these details...
- Your prefix to call out is 0 - I think it may be something else
- You are dialing the prefix before dialing the number....
- The number you are trying to call is 0614024168
** If you are trying to call 0614024168 then you need to put the prefix of the outbound rule in first... Mine is 0 but yours may be different...
** So the number to be dialed may be 0061402168

It doesn't look like 3CX is seeing that prefix and linking it to the outbound rule!...
To check your prefix go
OUTBOUND RULES > MANAGE
read the line column 'Number prefix'

If you are puting the prefix in then have you upgraded the system recently?
If so just delete the outbound rule and recreate it...
The outbound rules can become corupt during upgrades... This is stated during the upgrade process!

Post back how you go please!

 

[Patch]

I just like nanocomputers had 404 errors , changing port I get 503 errors.

I can't find my mistake by myself, after so many different configurations I never can dial out using pstn line.
I post all my important screenshots.

I must be doing some stupid error but... can't find it

 

[nanocomputers]

@ Jay,

Do you mean both the Linksys Gateway and 3CX must have a prefix? Where in the Linksys can I find the prefix settings?

thanks,

Gilbert

 

[nanocomputers]

I don't have a number prefix screen, this is the screen I see when I open the outbound rule I have now (see attachment)

 

[Moxi]

I hope this helps, try to config your 3CX and SPA3102 as I did.

3CX:
http://i32.tinypic.com/nyahlh.jpg
http://i29.tinypic.com/202yxy.jpg
http://i32.tinypic.com/2ia91xd.jpg
http://i30.tinypic.com/zodlid.jpg
SPA3102
http://i26.tinypic.com/2ibyg7c.jpg

Be sure that SPA3102 - Line 1 - SIP Port is not the same as PSTN SIP Port.

Disable NAT Keep Alive Enable because it constantly generates errors on the 3CX log (lots of MBytes in a day).

In your SIP Phone Dial 0 and you'll get PSTN tone, then dial the other numbers, or dial 0XXXXXXXXXXXXX to call directly.

(My Disconnect Tone is different because my SPA3102 is behind a PABX, otherwise it never hooks on)

 

[nanocomputers]

Thank you for your reply,

I changed everything to the way you have it, but still nothing happens. If I dial 0 for outside line it doesn't even give me a dial tone.
I think this really has got to be something with the Linksys, 3CX software is so easy and is setup like yours.

if wanted I can provide screenshots of my linksys spa3102, please tell me what screen you would like to see.

 

[Moxi]

3CX:
Set Logging Level to VERBOSE MODE at General/General Settings
Make a backup and then clear Logs Directory (C:\Program Files\3CX PhoneSystem\Data\Logs)
you will need to stop all 3CX Services. Windows:Administrative Tools/Services
Try to make some calls dialing 0123123123, 0321321321, 0.
Post your Logs directory.
SPA3102:
Post the Voice Info tab

Thank you for your reply,

I changed everything to the way you have it, but still nothing happens. If I dial 0 for outside line it doesn't even give me a dial tone.
I think this really has got to be something with the Linksys, 3CX software is so easy and is setup like yours.

if wanted I can provide screenshots of my linksys spa3102, please tell me what screen you would like to see.

 

[nanocomputers]

Here is the screenshot of the info tab of my Linksys SPA3102

 

[nanocomputers]

Here are the 3cx logs, hope someone can help me out...

 

[Moxi]

updated
Nanocomputers;

It seems like:
--Your 3CX is at 192.168.168.253
--Your SPA3102 is at 192.168.168.250
--Your IP Phone is using [email protected]:5060 but it seems that registers two times. My IP sofware phones just generate a line in the log while registering, I don't have one hardware IP phone so I can't compare.

Your log shows that:
Your 3CX initializing log is shorter, maybe you've disabled some unused services as i did.
Your SPA3102 is registering, your IP Phone is registering.

When you try to start a call with your IP Phone:
3CX receives the call and attempt to make a route it.
on next line log line it calls 192.168.1.250 and your SPA3102 doesn't answer the call, times out, so 3CX reports that route can't be reached.

I have searched for that Timeout error in my logs, I've found one and I think it happened when I was trying to call from a PC to a PocketPC whose wireless signal was to weak and was losing signal randomly. But I'm pretty sure that's not your case.

If your 3CX if behind a firewall check that the necessary RTP port range is allowed. Maybe just 5060 is allowed so SPA3102 is registering but can't communicate completely.
Try to configure a software IP phone to check if it works.

This is how I've made my SPA3102 to work:
1. I had a SIPPHONE account so i used following page:
http://voxilla.com/tools/device-configuration-wizard/linksys-spa3102/spa3000-configuration-wizard-806.html
Everything was OK.
2. Weeks later I got 3CX so I replaced the necessary parameters to use it. IPs, User IDs, Passwords, etc.
Everything is OK. Maybe step 1 configured something else in my SPA3102.

To SAVE/RESTORE your SPA3102 config:
http://forum.voxilla.com/linksys-spa9000-spa400-support-forum/save-backup-spa9000-config-25340.html

 

[nanocomputers]

Just to make sure:

I am not using voip to dial out, i am using a regular landline. Therefore I did not configure the firewall, I assume that this is correct.

I still think the problem is the Linksys like: PSTN and LINE are not working together... Weird thing though is that when I pick up the phone I hear a normal dial tone. When I call a number (with or without a prefix) it says "BUSY"

 

[adza]

Re:

Not in the proxy section.
PSTN Line tab -> SIP settings -> SIP port : 5060
Line1 tab SIP settings -> SIP port : 5061

sorry for my english

Stuff your English - it was perfect for my needs - this was a life saver!!! 8) Thanks heaps. Pitty after all this time the documentation hasn't been updated!

Cheers & thanks for your help!

Adam

 

[ddorry]

I just wanted to register and post my thanks to the people on this thread. Because of you, I was able to get this thing working after about 3 full days of searching and what not. The biggest thing for me was the switching of the ports for Line1 and PSTN. I believe the software revision 5.1.7 ( think ) has these as system defaults, but I changed them in my ignorance, thinking they had to be 5060 the first time. After reading this I did another system reset and followed the instructions and viola .. all is working and I understand it better.

Thanks again for taking the time to share your knowledge.

 

[galal202]

I just wanted to register and post my thanks to the people on this thread. Because of you, I was able to get this thing working after about 3 full days of searching and what not. The biggest thing for me was the switching of the ports for Line1 and PSTN. I believe the software revision 5.1.7 ( think ) has these as system defaults, but I changed them in my ignorance, thinking they had to be 5060 the first time. After reading this I did another system reset and followed the instructions and viola .. all is working and I understand it better.

Thanks again for taking the time to share your knowledge.

Hi ddorry,
You're welcome and I hope you enjoyed the 3CX as I am by

and you could see this topic also
http://www.3cx.com/forums/unable-to-configure-linksys-7530.html