SPA 3102 Outgoing

Discussion in '3CX Phone System - General' started by Piggy, Aug 13, 2007.

  1. Piggy

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    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::eek: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::eek:nIncoming [CM103002] Call(2): Incoming call from 101 (Ext.101) to sip:916265551212@192.168.2.101

    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::eek: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::eek:nProvisional [CM103003] Call(23): Ln:10008@PSTN-Gate is ringing
    22:30:53.282 CallConf::eek:nIncoming [CM103002] Call(23): Incoming call from 101 (Ext.101) to sip:916265551212@192.168.2.101

    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.
     
  2. Anonymous

    Anonymous Guest

    Are you trying to use the PSTN out via the SPA? or VSP?
     
  3. Piggy

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    I am doing PSTN out via SPA (3cx using SPA as gateway to dial out through normal POTS phone lines). I am also using 3cx Softphones as the phone client dialing out. I must be doing something wrong with that Line1 ain't I?
     
  4. renaissance

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    Try to change PSTN port to 5060 and Line1 port 5061
     
  5. Piggy

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    I am assuming you want me to change the Proxy section (e.g. 192.168.2.101:5061) to the ports you suggested. I've just tried that and it did not work. The PSTN is on port 5060 and Line1 port is 5061 otherwise.

    Also, The error I get on the 3cx Softphone is: 1:Unallocated (unassigned) number
     
  6. Piggy

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  7. renaissance

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

    sorry for my english
     
  8. Piggy

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    That works out great! At least I can call out. I'm experiencing the same problem as others where I can click "9" (to dial out), but have to wait for the dial tone first and then dial the number. I'll check with that thread for more info, unless someone knows the solution for that too.

    Thanks Renaissance!
     
  9. Mirzab

    Mirzab Member

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    I just had a couple of weeks of fun with this but it does work great now. Just on a brief read through, and I might be duplicating other advice already received, I would suggest the following:

    PSTN Gateway settings in 3CX: Set port default 5061; in Registration check "Use registrations contact field yada yada"; in Location of Destination number check "Request -Line-URL field".

    SPA PSTN Tab: Proxy 192.168.2.101 only, do not add port here, and it is already on 5061 in the "SIP Settings" section; in Dial Plan section clear Dial Plan 1 field and leave blank, do not change the other Dial plan settings; in FXO Timer section - PSTN Ans Delay = 3, PSTN ring thru delay = 1.5.

    Line 1 Tab in SPA - make sure on different port than PSTN, use 5060 here, simplest. Again make port changes in SIP Settings area and only IP address in Proxy field.

    Save and restart. ITFarmer suggests disconnect power after saving for a few seconds then plug in back, and he is very good, some of this came from him and was invaluable, the rest from Cjay (another Forum member).

    I am not sure what you are trying to do with Line 1 on the SPA, possibly trying to use it to dial out on PSTN? Never worked for me. I use it as an analog adapter for one of my internal extensions and works fine.

    I will setup my config pages and post on my server with links soon, got the idea from you above. If you wish copies sooner PM me with an email address and I will send.

    Mirza.
     
  10. Piggy

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    Thanks Mirza! Will try that once I am on location.

    Maybe I am confuse as I am confusing everyone with the Line1 purpose. Let me try to clarify what I am trying to do.

    What I'm trying to accomplish:
    I have 1 regular POTS phone (1 phone number). I used the spa 3102 with the 3cx to create a pbx system where incoming calls to the one phone number will be answered by the digital attendant and forward to proper extension. All outgoing calls simply uses the spa to dial-out like any phone (just like your regular analog phones). Its a very straight forward setup I believe.

    Confusion:
    With everything pretty much working (and satisfied at the moment)... So am I setting up the SPA properly? Am I not suppose to input any settings for Line1 on SPA? I just followed the 3cx Instructions on setting up 3cx. I really have no idea what Line1 is suppose to do, but I believe it allows outgoing calls to be established. Everyone seems to be confused about what I am trying to accomplish. I hope I clarify my goals and that I am walking the right path.
     
  11. Mirzab

    Mirzab Member

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    When it come to the SPA3102 simplicity and clarity apparently do not exist. There is a multitude of posts and documentation online everywhere that counters and conflicts each other in my personal opinion. There does appears to be a method to dial out PSTN via the Line 1 / VoIP side (I never got it to work) but I do not believe it is absolutely required. If all is working to your satisfaction right now, leave it as it is "If it ain't broke, don't fix it" - for myself Line 1 drives an analog extension allowing me to use a "traditional" phone set, and the PSTN fallback feature allows that phone to function in event of power failure longer that my UPS can sustain, or if the actual 3CX Server PC goes belly up. Otherwise, I don't believe its needed.
     
  12. Anonymous

    Anonymous Guest

    So you do not have a VSP just want to use the PSTN line and use the ATA to translate analogue to SIP. That is ok.

    Line 1 is required so it needs to be enabled to make PSTN calls and to receive PSTN calls. Once I knew why, but forgot (like so many things lately oops).

    Anyway, these are the specific settings to make PSTN only work:

    In the Line 1 Tab
    1. Dial plan enter something like <#9:><:mad:gw0>. #9 being what you dial to get through to the PSTN. You can use another combination if you like.

    2. In the PSTN Tab:
    Leave dial plan 1 empty
    Make sure VoIP-To-PSTN Gateway Enable is set to “Yes”
    Line 1 VoIP Caller DP: “1”
    VoIP Caller Auth Method: “PIN”
     
  13. Piggy

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    Thank you Rennaissance, Mirzab, and ITFarmer! I got everything to work just fine now! I even got a better understanding of this whole SPA-3CX thing from this topic thread. The suggestions from Rennaissance got the ip phone to connect to the PSTN line when I dial "9". Mirzab's post help me eliminate the need to press "9" and wait for a dial tone, then punch in the numbers. I can now punch in "91231234567" on the softphone or the Grandstream hardware and both can dial out and receive calls from regular phone lines just fine!! ITFarmers' post got me thinking on what Line1 really is. I disabled Line1 on the SPA while all the above was working like a charm. I was always confuse about what Line1 is (I may be over-thinking to something more complicated or I'm just stupid!), but after reading ITFarmer's post I realize its just a setting for the Phone jack on the SPA. Its like a built-in Analog Adapter (which you can buy for your analog phone). So I set that Line1 on the SPA up and plug in my cordless analog phone (with proper extensions and outbound rules set)... and everything worked!!!

    All of a sudden everything worked and everything made sense! Thanks alot guys! you guys save me weeks of headaches!!
     
  14. Mirzab

    Mirzab Member

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    Since you are using the single PSTN line outbound you can eliminate the "9" for external calls by working on the rules. Possibly create one for the local area code (mine would be something like all numbers beginning with 416) to use PSTN, then another for all numbers beginning with "1" since almost everyone dials "1" before a long distance area code. You don't need to delete the current rule for "9" while trying the new rules, and if all works well you would return to the same dialing method you had on the PSTN line directly before 3CX. Just a suggestion from someone with a knack of creating new headaches for himself, never learned to leave well alone :lol:
     
  15. Piggy

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    That's a great idea Mirzab! I didn't even think of it, I'll give that a try.
     
  16. Anonymous

    Anonymous Guest

    NO NO NO, step away from the SPA, I said step away from the SPA.......


    Oh I feel some more sleepless nights comming along.


    :lol: Just kidding :lol:

    Good to hear you are on your way.
     
  17. Mirzab

    Mirzab Member

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    Behave Henk! It works like a charm for me now, I just love it! Want a copy of the outbound rules I use? I really could write a manual on this thing now. :lol:
     
  18. DaveIW

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    WHOOHOO!!!!

    Thanks for this little snippet - this has just solved my problem with outgoing calls via 3cx and my SPA-3102.

    Pardon the expression, but I've been at it all day and chuffed to bits that both incoming and outgoing now works through 3CX.
     
  19. thefamilyman

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    hi, sorry to be a pain,

    I'm having similar difficulties.

    Basically what i want to do is make calls from my SIP phone out onto my land line.
    I've configured my cellphone with all the correct SIP settings, i can login to the 3CX server via my phone internet, i've tested to make calls out though Gizmo VoIP via 3CX this works fine, so that indicates to me that my SIP phone -> 3CX -> VoIP is working fine.
    BUT, i'm trying to configure the Linksys SPA for PSTN line out instead of the Gizmo VoIP. i want to go SIP phone -> 3CX -> PSTN.

    I've tried all your suggestions, well, atleast try to follow them, a little confusing atleast.

    generally what i have done is:
    set up an extention, with 10 as the ID, Name and passpord (to make it simple)
    The Outbound Rules are simple too, any call from extention 10 with the prefex of # is directed to "lineout" (this is what i've named the Linksys).

    I added a PSTN line on the 3CX with the ID and Password as 10000, with the port as 5060 (default)

    I've first configed the linksys with the help of the 3CX tutorial, this didn't work. For example, i want to call the number 8243766, on my sip phone i dial #8243766 (# to access the line out, this worked fine with Gizmo VoIP) but i get an error:
    Call::Terminate [CM503008]: Call(1): Call is terminated
    Call::RouteFailed [CM503014]: Call(1): Attempt to reach [sip:%238243766@192.168.0.91] failed. Reason: Not Found
    CallLeg::eek:nFailure [CM503003]: Call(1): Call to sip:98243766@192.168.0.100:5060 has failed; Cause: 404 Not Found; from IP:192.168.0.100
    CallCtrl::eek:nSelectRouteReq [CM503004]: Call(1): Calling: PSTNline:10000@[Dev:sip:10000@192.168.0.100:5061]
    CallCtrl::eek:nIncomingCall [CM503001]: Call(1): Incoming call from Ext.10 to [sip:%238243766@192.168.0.91]
    ExtnCfg::updateContact [CM504001]: Ext.10: new contact is registered. Contact(s): [sip:10@192.168.0.84;transport=UDP/10]

    so i noticed it was trying to use port 5061 (dunno why) so i changed the port on Line1 tab on the SPA to 5060, this seemed to change the error:
    Call::Terminate [CM503008]: Call(19): Call is terminated
    Call::RouteFailed [CM503014]: Call(19): Attempt to reach [sip:%238243766@192.168.0.91] failed. Reason: Server Failure
    CallLeg::eek:nFailure [CM503003]: Call(19): Call to sip:8243766@192.168.0.100:5060 has failed; Cause: 503 Service Unavailable; from IP:192.168.0.100
    CallCtrl::eek:nSelectRouteReq [CM503004]: Call(19): Calling: PSTNline:10000@[Dev:sip:10000@192.168.0.100:5060]
    CallCtrl::eek:nIncomingCall [CM503001]: Call(19): Incoming call from Ext.10 to [sip:%238243766@192.168.0.91]

    I've tried a combination of all your suggestions, it seems to be swapping between those errors, to do with the ports depending on what the errors are.

    So i'm asking a really noob request;
    Can someone please make a little sense of these suggestion and settings and make a simple "Setup Guide" to make calls out from and extension to a landline via the Linksys.

    I hope i make some sense out of this hehe.
    I just would love some help and i feel it would also be greatly beneficial to any new user of the Linksys as its rather unfriendly to setup.

    Many thanks to those who can help!!!!
     
  20. nanocomputers

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    Hi,

    Hopefully someone can help me out here, because this is giving me headaches at the moment. Trying to get the outside line working for over a week now.

    I am using 3cx, with the Linksys SPA-3102 and a Tiptel 220 Voip.

    Incomming calls from the normal landline is working great, even with Caller ID.
    Outgoing calls always needs to go over the landline (pots) because I only dial landline numbers which are free of charge for me.

    Despite all the good options given in this post, I cant manage to get it working right.

    This is the log when I try to call out:

    22:43:06.609 Call::Terminate [CM503008]: Call(85): Call is terminated
    22:43:06.593 Call::RouteFailed [CM503014]: Call(85): Attempt to reach [sip:0614024168@192.168.168.253] failed. Reason: Not Found
    22:43:06.593 CallCtrl::eek:nSelectRouteReq [CM503013]: Call(85): No known route to target: [sip:0614024168@192.168.168.253]
    22:43:06.546 CallCtrl::eek:nIncomingCall [CM503001]: Call(85): Incoming call from Ext.1000 to [sip:0614024168@192.168.168.253]

    No known route to target, looks like a small problem to me... unfortunately I am not able to solve it.

    If someone needs more info just tell me what you need and I will post it.

    I am in Holland where all phonenumbers start with a 0, in 3cx the extension and the line or ok (green).
     

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