Help with PSTN caller ID presentation to SIP client

Discussion in '3CX Phone System - General' started by Borsoock, May 22, 2007.

  1. Borsoock

    Borsoock New Member

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    Hello
    OK, I give up... For last 2 hours I've been trying to set up Linksys SPA-3102 and 3CX for sending PSTN Caller ID to a SIP client. Unfortunatelly all I can see on the SIP phone is gateway ID or gateway IP address, no CID from PSTN. Maybe I'm missing something so, can somebody help me with appropriate settings please?

    Thanks in advance

    Marcin
     
  2. archie

    archie Well-Known Member
    3CX Staff

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    Hi,

    Can you make a wireshark (or ethereal) capture of incoming call from PSTN gateway? Than I can try to answer your question, or make necessary fixes for next build.
     
  3. Borsoock

    Borsoock New Member

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    Thanks Archie.

    I'll prepare dump, but still please advice wchich gateway advanced settings combination suppose to be correct for this purpose.

    Thanks

    Marcin
     
  4. archie

    archie Well-Known Member
    3CX Staff

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    Oh, it's a difficult question for me, because i've never tried to set it up. I know that ITFarmer has prepared detailed manulal on setting up that device. I believe it was posted on the forum somewhere. Or PM him for your question.
     
  5. Costas3CX

    Costas3CX New Member

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    I have prepared one too recently. but found a small error in it and have to update the faq in the website. Indeed IT farmer has created one too way before anyone of us even tried to do such a thing.

    Just remember that if you are going to use both the fxo and fxs interfaces that they have to be configured on different ports (which I forgot to mention in my faq and thats why i have to update it).
    http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE
     
  6. Borsoock

    Borsoock New Member

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    Hi Costas
    I use FXO only. With settings from ITfarmer's website I can see only gateway IP address on SIP clients (tested with x-lite and Avaya 6425). Still no PSTN CID forward :( . I'll prepare ethereal dump for you in the near future.

    Thanks

    Marcin
     
  7. Raja Adil

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    I have struggled a lot to get the PSTN caller id but in the end i got it i.e the telephone number of the caller. If you have configured the linksys 3102 gateway and able to inbound and outbound calls as you told you are getting the gateway ID or IP. If you are getting ID of the gateway that would be something like 103 which is in fact extention you have created. Please follow these steps .

    Step 1. Open linksys 3102 web administrator. Click on Admin Login and then click advance now click on PSTN. Under PSTN-To-VoIP Gateway Setup you will see PSTN CID For VoIP CID: if it is no make it yes.

    Step 2. Now logon to 3CX and click on Server Status. You will see the activity of the 3CX. Now dial your gateway line, Please dont worry if you are not getting the in-coming call. No try to concentrate on the activity log. It will give you some error trace like give under. Here is key to door that we will try to get the caller ID of the in coming call from PSTN. Look in contact=0515598889@192.168.0.230:5061 I have 0515598889 as my caller number and 192.168.0.230 is ip of linksys gateway and 5061 is the port of the linksys 3102.

    CM102001]: Authentication failed for SipReq: INVITE 106@192.168.0.155 tid=-9e0a8575 cseq=INVITE contact=0515598889@192.168.0.230:5061 / 102 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
    10:00:05.988 evt::CheckAuth::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
    INVITE sip:106@192.168.0.155 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.230:5061;branch=z9hG4bK-9e0a8575
    Max-Forwards: 70
    Contact: "LinkSys103"[sip:0515598889@192.168.0.230:5061]
    To: [sip:106@192.168.0.155]
    From: "LinkSys103"[sip:0515598889@192.168.0.155];tag=598aa9c3a8cc429o1
    Call-ID: c2d6c54b-dde01e41@192.168.0.230
    CSeq: 102 INVITE
    Expires: 240
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Proxy-Authorization: Digest username="103",realm="3CXPhoneSystem",nonce="12851413205:3a86fb7719baeb097c6424f7a8d4a17b",uri="sip:106@192.168.0.155",algorithm=MD5,response="d89a28e2e8a8b4de597d5be0e207b811"
    Supported: x-sipura
    User-Agent: Linksys/SPA3102-3.3.6(GW)
    Content-Length: 0
    Remote-Party-ID: LinkSys103 [sip:0515598889@192.168.0.155];screen=yes;party=calling
    Diversion: Gateway10001 [sip:10001@192.168.0.155];reason=unconditional


    Step 3. This error log given above is generated as 3CX assumes 0515598889 as sip id from the statement "sip:0515598889@192.168.0.155" (taken from the above log). We have to Authenticate it in 3CX. In step 3 remove password from the gateway. To do this click manage in Lines in the 3CX management console. Now you will see you gateway some thing like 10000 ... click on Gateway/Provider Name, In my case it is LinksysGateway it is name given to my gateway when gateway was added. You will find Source Identification here.

    there will be three rows under it.
    You have to set only first row, In the SIP Field: choose-> Contact: Host Part
    In Value choose -> Custom Field
    Custom Value -> 192.168.0.230:5061

    Now dial the call I hope you will get the Caller ID i.e. hhe telephone number of the caller.
     
  8. negoceb

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    Brilliant!! Thanks for share!
     

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