3CX System and SPA8800 gateway

Discussion in '3CX Phone System - General' started by dan.dinica, Apr 15, 2013.

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  1. dan.dinica

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    Hi,

    I try to setup the following configuration :
    - 3CX Phone System software ( ver. 10 or 11, it is not important ) free edition.
    - 10 SPA 502G phones
    - 1 x SPA 8800 for our analog lines ( 2 ) and one line from a GSM adapter .

    Until now ( I do not have the physical devices until now but it will be soon ) I setup a configuration with few extensions and I made some tests with softphones and android devices and everything seems to be ok; I made tests with the phones from the inside network and also from the outside . Now I want to make the next step, e.g. to configure the gateway and the analog lines ( 2 ) , and no need for a SIP provider in this moment. When I add a new gate to the system ( my SPA 8800 ), the only option I have is to select for the brand = Generic and for the model = Gateway Device . After, I must choose the number of ports and I have the first question : I must fill 4 or 8 ?( my opinion is 4 but.... ) When I goes to the Create Outbound Call Rule page , the second question comes : how can I setup a route in 3CX software, for an outbound call based on rules I setup here ? ( e.g. if I want some outbound calls to goes to the analog lines, ports 1 and 2 from the gateway , and other calls for mobiles phones to goes on the third port thru the GSM adapter ). It is the SPA 8800 who make the difference between calls based on the rules ? ( I do not have yet the physical device and I do not see many posts on forum about this subject ).
    Thanks in advance for any suggestions .
     
  2. lneblett

    lneblett Well-Known Member

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    There are a couple of things missing in your set-up.

    1. The SPA8800 is an 8 port device supporting 4 FXO and 4 FXS devices. You will be using the FXO side of the device to support the PSTN lines, so I am not sure what the intent is with the FXS lines given that you also bought 10 IP phones. This might be overkill for your set-up, but so be it.

    2. You do not mention the make/model of the GSM gateway.

    My best advise is that your should try and limit your hardware selections to those that are specifically mentioned as preferred or supported by 3CX. You'll find more support amongst the community as the familiarity will be better and you also help to ensure future compatibility.

    in any event, as you integrate the devices into 3CX each will be represented as a separate device and given a name assigned by you. It matters not what the name is, but rather the name will be used to decide routing. Each device will also be associated a port by 3CX based upon how you program the device and its physical characteristics. In your case. I suppose you would program the SPA to have the number of ports in use based upon what you expect the FXO connectivity to ultimately be (no more than 4). You would then program the GSM gateway accordingly and as such, there are now 2 gateways attached to the system.

    Once the devices are provisioned, you then establish the outbound rules using the criteria you want that directs calls to either the SPA or to the GSM device....say dialing a "9" will send to the GSM whereas all other calls will go to the SPA. You may be able to use logic that is already contained with the intended dial string to decide which path to take.
     
  3. dan.dinica

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    Thanks for the quick answer ,
    If I well understand the description of what I want to do was a little bit incomplete and I will givw you more details :
    - from the SPA 8800 I will use 2 FXO ports for the 2 analog line from the telephone company ( ports 1 and 2 )
    - on the third port ( port 3 ) I will plug the GSM adapter who work also like an analog line ( e. g. like an FXS port )
    - the last FXO port of the SPA 8800 can be disabled in this moment .
    - on the FXS ports I intend to plug maybe 1 or 2 analog phones and a fax device ( !!! ) - this is my wish but I will see if it is possible,
    - having this configuration I want to setup the IP-PBX to route the calls for local fixed posts to ports 1 and 2 and the calls for mobile users on the port 3,
    - the question is where and how I can write one or 2 - 3 rules to make the software to take the wright decision with the calls ( from the software I see only the rules for the format of called numbers and not how they will quit the local network ).
    What is your opinion related to my description ?
     
  4. leejor

    leejor Well-Known Member

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    To separate local/landline calls from Mobile calls (easily), requires that you live in a country where the mobile number has a specific (unique) prefix. Outside of North America, that is usually the case.

    In England, for example, mobile number begin with 07, so you could have an outbound rule that sent all dialled numbers beginning with 07, and were 11 digits long, out on the GSM gateway. If you put that rule ahead of a "catch all" rule to deal with all other 0X numbers, then all 11 digit number NOT beginning with 07, could be sent out on the other two PSTN lines.

    You will also probably need additional rules to deal with other "special" (short) length, or local numbers.

    It will all depend on the dialling plan in your area/country.
     
  5. eagle2

    eagle2 Well-Known Member

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    You better treat SPA8800 as 4 separate SPA3102s, they share the same IP address, but different ports.

    BR
     
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  6. dan.dinica

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    Hi Eagle2,

    I thought also at this little trick but I did not know about the shering of the IP address .! There is also an another problem , reading some documents related to SPA 3102 I saw it is not possible to use a fax device - it not function like PSTN gateway and ATA device in same time , I am wright ( see the link : http ://www.3cx.com/voip-gateways/linksys-3102/) - NO LONGER AVAILABLE ??
    Bye the way can somebody to give me a link where I can found detailed description ( software manual )on how to setup and configure the SPA504G and SPA8800 ( all the manuals I found does not explain piece by piece the menu from the web admin , only give some simple indications about things who are not so important ). :)
     
  7. leejor

    leejor Well-Known Member

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    Fax through a gateway/ATA requires that both support t.38. Some Linksys/Cisco devices have had firmware upgrades that allow this, you'll have to go through the settings of your device to confirm this.

    I realize that the 3CX device documentation mentions that the use of both the ATA and gateway (at the same time),on the 3102, is not supported. that may be that case, but I've not had a problem getting it to work. That doesn't mean that you won['t encounter any problems using both the gateways and ATA's on the 8800, but, I would tend to think that it will work for you.

    If you can't find a link to a manual for the 8800, here is a link to some other Linksys manuals, most of the terms and settings are common to many of their devices.

    http://www.belldandy.idv.hk/wordpress/i ... s-ata-pdf/

    One of the problems finding a step by step set-up guide for use with 3CX (other than perhaps on the 3CX site), is that the use and settings can vary greatly from site to site.
     
  8. eagle2

    eagle2 Well-Known Member

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    Check the Cisco site or google for admin guide.

    SPA3102 may be used in many ways (at least 7 scenarios), generally you may thin of it as two devices - one device with 1 FXS port and another (independent) device with 1 FXO port.

    You can't provision both ports simultaneously, but is fairly easy to configure them manually, I even prefer it (check 3CX guidelines for manual configuration of SPA devices / phones). There are some important details for configuring the FXO port.

    There are lots of posts on SPA3102 and other SPA devices, 8800 is simply 4 x 3102s put into one box, sharing one IP address (8 different ports).

    BR
     
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  9. dan.dinica

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    Hello all,


    Can someone to help me to understand where is the problem with the following log messages :
    ---- from Server activity log ----

    13:53:19.875 [CM503020]: Normal call termination. Reason: Not found
    13:53:19.875 [CM503016]: Call(192): Attempt to reach <sip:00731304662@10.100.1.8> failed. Reason: Not Found
    13:53:19.875 [CM503003]: Call(192): Call to sip:00731304662@10.100.1.90:5060 has failed; Cause: 404 Not Found; from IP:10.100.1.90:5060
    13:53:19.765 [CM503025]: Call(192): Calling PSTNline:00731304662@(Ln.10000@Linii analogice)@[Dev:sip:10002@10.100.1.90:5261]13:53:19.750 [MS210002] C:192.4:Offer provided. Connection(transcoding mode): 10.100.1.8:7366(7367)
    13:53:19.734 [CM503003]: Call(192): Call to sip:00731304662@10.100.1.90:5060 has failed; Cause: 404 Not Found; from IP:10.100.1.90:5060
    13:53:19.671 [CM503025]: Call(192): Calling PSTNline:00731304662@(Ln.10000@Linii analogice)@[Dev:sip:10001@10.100.1.90:5161]
    13:53:19.671 [MS210002] C:192.3:Offer provided. Connection(transcoding mode): 10.100.1.8:7364(7365)
    13:53:19.625 [CM503003]: Call(192): Call to sip:00731304662@10.100.1.90:5060 has failed; Cause: 404 Not Found; from IP:10.100.1.90:5060
    13:53:19.562 [CM503025]: Call(192): Calling PSTNline:00731304662@(Ln.10000@Linii analogice)@[Dev:sip:10000@10.100.1.90:5061]
    13:53:19.562 [MS210002] C:192.2:Offer provided. Connection(transcoding mode): 10.100.1.8:7362(7363)
    13:53:19.531 [CM503004]: Call(192): Route 1: PSTNline:00731304662@(Ln.10000@Linii analogice)@[Dev:sip:10000@10.100.1.90:5061,Dev:sip:10001@10.100.1.90:5161,Dev:sip:10002@10.100.1.90:5261]
    13:53:19.531 [CM503010]: Making route(s) to <sip:00731304662@10.100.1.8>
    13:53:19.531 [MS210000] C:192.1:Offer received. RTP connection: 10.100.1.90:16428(16429)
    13:53:19.531 Remote SDP is set for legC:192.1
    13:53:19.531 [CM505001]: Ext.125: Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA8800-6.1.7(GW)] PBX contact: [sip:125@10.100.1.8:5060]
    13:53:19.515 [CM503001]: Call(192): Incoming call from Ext.125 to <sip:00731304662@10.100.1.8>
    13:53:19.484 [CM500002]: Info on incoming INVITE:
    INVITE sip:00731304662@10.100.1.8 SIP/2.0
    Via: SIP/2.0/UDP 10.100.1.90:5060;branch=z9hG4bK-e7b0a702
    Max-Forwards: 70
    Contact: "Fax"<sip:125@10.100.1.90:5060>
    To: <sip:00731304662@10.100.1.8>
    From: "Fax"<sip:125@10.100.1.8>;tag=78c555c944b23cb6o0
    Call-ID: ae039d85-a9b4382a@10.100.1.90
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="125",realm="3CXPhoneSystem",nonce="414d535c0781b31f43:b03229047b19d196da989e71128e3c0d",uri="sip:00731304662@10.100.1.8",algorithm=MD5,response="d7b0dc06917137602e640a74e6488551"
    Supported: x-sipura, replaces
    User-Agent: Cisco/SPA8800-6.1.7(GW)
    Content-Length: 0
    Remote-Party-ID: "Fax" <sip:125@10.100.1.8>;screen=yes;party=calling

    ------ and from Server Event log -------

    Call to 00731304662@(Ln.10000@Linii analogice) has failed. [ V4 10.100.1.90:5060 UDP target domain=unspecified mFlowKey=768 ] replied: 404 Not Found; from IP:10.100.1.90:506
    -------------------------------------------------------------------------------

    The logs represent a test to call an outside number on a PSTN line from the SPA 8800. The port 10000 is the Line 1 ( FXO ) from the gateway SPA8800 ( with the name Linii analogice ) with the IP 10.100.1.90, and the server 3CX has IP add. 10.100.1.8 . This PSTN line is for the moment an extension line from our analog PBX , and for going outside we need a "0" in front of the number e.g. if the number is 0731xxxxxx we must dial 00731xxxxxx . 125 is the extension for test .
    I can say also I can call from outside ( from our analog PBX ) an extension from 3CX .
    I do not understand why SPA 8800 can not put a call to this line .

    Thanks in advance,
     
  10. eagle2

    eagle2 Well-Known Member

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    It is much simpler,

    send me a private message with your e-mail / phone / skype, I could help you setting it (using Teamviewer, Ammyy).
    I need also to have an idea of your setup (IP address of PBX, SPA device, etc.).

    BR
     
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  11. leejor

    leejor Well-Known Member

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    Check your port assignments/settings. I'm not sure if you meant to use 50XX, 51XX and 52XX

    You could use 5060, 5061, 5062 and 5063 for the four ATA devices. They will register on their own and don't require any corresponding port number set in 3CX.

    You can then use 5064, 5065, 5066 and 5067 for the four Gateway ports. These port numbers must be set in each of the four the 3CX trunk settings and match up with the gateway ports.
     
  12. eagle2

    eagle2 Well-Known Member

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    Default ports in SPA8800 are 5060&5061, 5160&5161, 5260&5261 and 5360&5361 for each of the SPA-3102s inside.
    Ports 5x60 are used for FXS, ports 5x61 are FXO. Of course you can change the ports, if you like, but this is not the issue of Dan (I think the problem is related to provisioning and inability to provision FXS and FXO port (of SPA-3102) independently, moreover 3CX blog's manual configuration guidelines is not intended to use the FXS and FXO ports independently).

    I would recommend using separate SPA-3102s instead, moreover this will be cheaper.

    BR
     
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  13. dan.dinica

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    Hello my friends,
    I am back again, with a new question. I solved the 404 error and now I can make outside calls to PSTN lines tied to a SPA 8800 and also to use analog phone on the the FXS ports. The only challange I will have with this device is to make the fax machine to work on it.
    My new question is related to the following problem :
    - My 3 analogues lines are pluged in the SPA 8800 and the virtual ports asigned by the 3CX software are 10000, 10001 and 10002 .
    How ( or there is any possibilities ) I can monitor these lines ( ports ) with an SPA 504 phone from Cisco who has 4 lines with LED light on each line . In my scenario I hope I can use one line for the operator extension and 3 BLF ( lights-LED ) for monitoring these lines for in and out calls. I want to monitor these lines because I want the operator to know what is the state of the lines in case of an urgent problem, and also to know how to answer multiple calls ...!

    Regards,
     
  14. leejor

    leejor Well-Known Member

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    This has been brought up in the past, and there have been suggestions that it is possible. The assumption (I gather) is that you would treat the trunk numbers as extension numbers in the BLF key set-up. I don't recall seeing confirmation that anyone has yet got it to work however.

    Even if both devices, the gateway and the ATA attached to the fax machine, are capable of t.38, it can still be a bit of trial and error to get fax working reliably over VoIP.
     
  15. dan.dinica

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    Hello friends,

    I tray to use my SPA-8800 to send fax, but something does not work well . Even in logs from 3CX system I do not find any error the process finish in a wrong way:

    14:05:48.778 [CM503008]: Call(21): Call is terminated
    14:05:48.637 Remote SDP is set for legC:21.1
    14:05:47.512 Session 151810 of leg C:21.1 is confirmed
    14:05:47.434 [CM503007]: Call(21): Device joined: sip:10000@10.100.1.90:5061
    14:05:47.418 [CM503007]: Call(21): Device joined: sip:129@10.100.1.90:5360
    14:05:47.403 [MS210007] C:21.1:Answer provided. Connection(by pass mode): 10.100.1.90:16443(16444)
    14:05:47.309 [MS210001] C:21.2:Answer received. RTP connection[unsecure]: 10.100.1.90:16443(16444)
    14:05:47.309 Remote SDP is set for legC:21.2
    14:05:47.309 [CM505002]: Gateway:[RomTel1] Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA8800-6.1.7(GW)] PBX contact: [sip:10000@10.100.1.8:5060]
    14:05:47.309 [CM503002]: Call(21): Alerting sip:10000@10.100.1.90:5061
    14:05:47.231 [CM503025]: Call(21): Calling PSTNline:0251470501@(Ln.10000@RomTel1)@[Dev:sip:10000@10.100.1.90:5061]
    14:05:47.231 [MS210006] C:21.2:Offer provided. Connection(by pass mode): 10.100.1.90:19398(19399)
    14:05:47.200 [CM503004]: Call(21): Route 2: PSTNline:0251470501@(Ln.10001@Romtel2)@[Dev:sip:10001@10.100.1.90:5161]
    14:05:47.200 [CM503004]: Call(21): Route 1: PSTNline:0251470501@(Ln.10000@RomTel1)@[Dev:sip:10000@10.100.1.90:5061]
    14:05:47.200 [CM503010]: Making route(s) to <sip:00251470501@10.100.1.8>
    14:05:47.200 [MS210000] C:21.1:Offer received. RTP connection: 10.100.1.90:19398(19399)
    14:05:47.200 Remote SDP is set for legC:21.1
    14:05:47.200 [CM505001]: Ext.129: Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA8800-6.1.7(GW)] PBX contact: [sip:129@10.100.1.8:5060]
    14:05:47.184 [CM503001]: Call(21): Incoming call from Ext.129 to <sip:00251470501@10.100.1.8>
    14:05:47.153 [CM500002]: Info on incoming INVITE:
    INVITE sip:00251470501@10.100.1.8 SIP/2.0
    Via: SIP/2.0/UDP 10.100.1.90:5360;branch=z9hG4bK-ddb893fb;rport=5360
    Max-Forwards: 70
    Contact: "Fax"<sip:129@10.100.1.90:5360>
    To: <sip:00251470501@10.100.1.8>
    From: "Fax"<sip:129@10.100.1.8>;tag=3ea5fe4c2dae88fo0
    Call-ID: c9eb8e48-9122ed7b @192.168.0.4
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="129",realm="3CXPhoneSystem",nonce="414d535c079c140b04:690e58776c9aa639bf89107b97963dc5",uri="sip:00251470501@10.100.1.8",algorithm=MD5,response="fd53e1b0b5358a1fdaaf342cf43d507e"
    Supported: x-sipura, replaces
    User-Agent: Cisco/SPA8800-6.1.7(GW)
    Content-Length: 0
    Remote-Party-ID: "Fax" <sip:129@10.100.1.8>;screen=yes;party=calling
    -------------------------------------------------------------------------------------

    My testing configuration is the following :
    - 2 analog lines ( PSTN lines ) are pluged in my SPA 8800 lines ports and are used to communicate with the outside world.
    - my fax machine is plugged in one of the ports from the ATA side of the gateway ( extension 129 ).
    - from my fax I can call any extension from inside network and also I can call someone from inside using one of the outside PSTN lines ( e.g. the call goes from fax to proxy server to the SPA 8800, in the public telephone company on one of the lines and back inside my company using the other PSTN line).
    - I have tried the same scenario with the fax as you can see from the log. but the message stop somewhere in the point when it tray to goes outside my company through SPA-8800.
    - I used for configuration of the SPA 8800 the notes from the following link :
    http: //www.3cx.com/sip-phones/linksys-spa-3102-fax/ - NO LONGER AVAILABLE. NEW LINK http://www.3cx.com/sip-phones/
    Can somebody give me some advice related to that configuration and/or where can be the error ?

    Thanks in advance and best regards
     
  16. dan.dinica

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    Hello my friends,

    Can somebody to help me to configure an SPA 8800 to have caller ID ( CID ) for a Vodafone line ( I use an adapter to have access in Vodafone network who have an FXS port who is plugged in my SPA 8800 in a line port ) ??

    :oops:
     
  17. eagle2

    eagle2 Well-Known Member

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    Set an answer delay of 3 seconds or so for calls arriving on your FXO port (LINE).
     
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  18. dan.dinica

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    Hi Eagle2,

    I have made this setting few days ago but without any succes. For the configuration of the FXO ( line ) from SPA 8800 I have used the article from the blog : Configuring Linksys 3102 for 3CX Phone System. For me the notes from step 20 seems to be the problem, because in my opinion the dial plan ( S0<:xxxxx>) means : what was dialed will be replaced with xxxxx where xxxxx is the port number for the PSTN line, and this was made to tell to the 3CX server where the incoming call must go, I am right ?? ( please tell me if I am wrong !!! ) . From the logs I can only see :

    22-May-2013 21:14:04.996 [CM503001]: Call(C:60): Incoming call from Line:10002<<10002 to <sip:800@10.100.1.10:5060>
    22-May-2013 21:14:04.996 NAT/ALG check:L:60.1[Line:10002<<10002] REQUEST 'INVITE' - basic check passed. No information for extended checks

    In conclusion how will be possible to see this CID even if in the start it will not be correct because of the chosen method ( Bellcore, DTMF....).

    Regards
     
  19. dan.dinica

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    Hi again Eagle2,

    Today making some tests I see the CID for my mobile number displayed next to the corresponding line in 3CX console, don't ask why because I don't no !!! After few tests the conclusion is : 1 of 2 outside calls generate CID ( in 95% of cases ) , and when is generated it is a good one . Have you any idea about it , can be a timing problem, or wrong method or a wrong CID FSK standard ??
    I have changed the PSTN answer time to 4 sec. , I have changed the method from Bellcore to ETSI FSK, also the Caller ID FSK
    Standard from bell 202 to v23, but the same result, in almost cases after 1 good test a bad one ( the ID of the port instead the CID ) is present !


    Regards
     
  20. leejor

    leejor Well-Known Member

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    Of course, you have to have the correct settings (CID type) or the trunk number will be sent. If you are using Vodafone, the i have to assume that it uses standard UK type caller ID. That should be ETSI FSK with PR (this may be "labelled differently depending on firmware version).

    Here is a link to a site that deals with setting up the 3102 for the UK, most setting should apply to the 8000. Keep track of what you change, and don't change anything just for the sake of changing it, it might not have the desired affect.

    http://www.aoakley.com/articles/2008-01-08.php
     
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