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Inbound caller ID "LASTNAME" gets stripped by Sangoma 101

This is a discussion on Inbound caller ID "LASTNAME" gets stripped by Sangoma 101 within the Sangoma forums, part of the Free VoIP Technical Support category; We have 3CX v10 installed with Sangoma 101 Pri card. Inbound CID are being SNIPPED by sangoma PRI card as ...

  1. #1
    sanyaolu is offline Junior Member
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    Default Inbound caller ID "LASTNAME" gets stripped by Sangoma 101

    We have 3CX v10 installed with Sangoma 101 Pri card. Inbound CID are being SNIPPED by sangoma PRI card as below. How can i resolve this ?

    The log shows
    From: "ROB" sip:6137xxxxxxx@10.146.7.2:5066;

    Instead of
    From: "ROB SMITH" sip:6137xxxxxxx@10.146.7.2:5066;

  2. #2
    Matthew's Avatar
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    Is this from the 3CX log?

    There isnt a Caller ID First and last name field that could be stripped, they are both the same field. Are you sure Rob Smith is what is reaching the Sangoma? Do you have any logs showing the proper caller ID?
    Matthew Hanson
    888Voip Systems Engineer
    www.888voip.com



  3. #3
    sanyaolu is offline Junior Member
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    The same PRI line is used on the prior PBX and it does pass First Name and Last name. I had a case with 3CX and 3CX support said that 3CX passes the CID as it received it from Sangoma.

    So if PRI works okay with prior PBX and 3CX said they pass the CID as they received it from Sangoma. This leaves me to look further on Sangoma.

    Please advice

  4. #4
    Matthew's Avatar
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    I understand this. But I think what you are missing is that the Netborder cannot "clip" a small part of the caller ID. It is one solid field. And it passes on to 3CX. It is not if it is ignoring a field, or something of that nature.

    Do this:

    In the Gateway Manager, fo to Satus and Controls => System Status, and check for errors. Post any Errors.

    Also, try running a Wireshark capture of the event. Check the traffic for the Sangoma to 3CX exchange, and note the caller name listed there. Post the SIP traffic messages.
    Matthew Hanson
    888Voip Systems Engineer
    www.888voip.com



  5. #5
    sanyaolu is offline Junior Member
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    INVITE sip:2221@10.146.7.135:5060;intercom=true SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-7925691d8d6c6f68-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:MakeCall@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;intercom=true
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 1 INVITE
    Alert-Info: <http://www.notused.com>;info=ivr
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Call-Info: <sip:3cx.pbx>;answer-after=0
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 403

    v=0
    o=3cxPS 283132297216 237632487425 IN IP4 10.146.7.2
    s=3cxPS Audio call
    c=IN IP4 10.146.7.2
    t=0 0
    m=audio 7284 RTP/AVP 0 8 3 13 9 110 99 101
    c=IN IP4 10.146.7.2
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:13 CN/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:110 iLBC/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    SIP/2.0 100 Trying
    To: <sip:2221@10.146.7.2:5060>;intercom=true
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 1 INVITE
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-7925691d8d6c6f68-1---d8754z-
    Server: Linksys/SPA962-5.1.15(aSC)
    Content-Length: 0

    SIP/2.0 200 OK
    To: <sip:2221@10.146.7.2:5060>;intercom=true;tag=999cf d1adf2e81di0
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 1 INVITE
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-7925691d8d6c6f68-1---d8754z-
    Contact: "Recpt Receptions" <sip:2221@10.146.7.135:5060>
    Server: Linksys/SPA962-5.1.15(aSC)
    Content-Length: 208
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: replaces
    Content-Type: application/sdp

    v=0
    o=- 4074299 4074299 IN IP4 10.146.7.135
    s=-
    c=IN IP4 10.146.7.135
    t=0 0
    m=audio 16386 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    ACK sip:2221@10.146.7.135:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-d422e01ded246113-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:MakeCall@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;tag=999cfd1adf2e81di0;i ntercom=true
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 1 ACK
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 0

    INVITE sip:2221@10.146.7.135:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-3640907d8e45c326-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:MakeCall@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;tag=999cfd1adf2e81di0;i ntercom=true
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 2 INVITE
    Alert-Info: <http://www.notused.com>;info=ivr
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Call-Info: <sip:3cx.pbx>;answer-after=0
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 421

    v=0
    o=3cxPS 270700380160 114991038465 IN IP4 10.146.7.2
    s=3cxPS Audio call
    c=IN IP4 10.146.7.2
    t=0 0
    m=audio 7284 RTP/AVP 0 8 3 13 9 18 110 99 101
    c=IN IP4 10.146.7.2
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:13 CN/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:110 iLBC/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=sendonly
    SIP/2.0 200 OK
    To: <sip:2221@10.146.7.2:5060>;tag=999cfd1adf2e81di0;i ntercom=true
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 2 INVITE
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-3640907d8e45c326-1---d8754z-
    Contact: "Recpt Receptions" <sip:2221@10.146.7.135:5060>
    Server: Linksys/SPA962-5.1.15(aSC)
    Content-Length: 208
    Content-Type: application/sdp

    v=0
    o=- 4074299 4074300 IN IP4 10.146.7.135
    s=-
    c=IN IP4 10.146.7.135
    t=0 0
    m=audio 16386 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=recvonly
    ACK sip:2221@10.146.7.135:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-9a610218db6bab19-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:MakeCall@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;tag=999cfd1adf2e81di0;i ntercom=true
    From: "MakeCall"<sip:MakeCall@10.146.7.2:5060;nf=v>;tag= 2728be1e
    Call-ID: YWQxZjRjODA0ZjIyZDFiNTJmYTNiMTQwMWNjZWI5Y2U.
    CSeq: 2 ACK
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 0

    INVITE sip:2221@10.146.7.135:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-2a0eb676d5073411-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:9050000000@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;tag=84cdf53da5a92854i0; intercom=true
    From: "Recpt Receptions"<sip:9050000000@10.146.7.2:5060;nf=v>;t ag=586ff962
    Call-ID: NTFhMGQ2YzQ3N2M0ZjI1ZWNhMWExM2Q3YmY4ZjMyOGU.
    CSeq: 4 INVITE
    Alert-Info: <http://www.notused.com>;info=ivr
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Call-Info: <sip:3cx.pbx>;answer-after=0
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 421

    v=0
    o=3cxPS 111333605376 398458880003 IN IP4 10.146.7.2
    s=3cxPS Audio call
    c=IN IP4 10.146.7.2
    t=0 0
    m=audio 7294 RTP/AVP 0 8 3 13 9 18 110 99 101
    c=IN IP4 10.146.7.2
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:13 CN/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:110 iLBC/8000
    a=rtpmap:99 SPEEX/8000
    a=rtpmap:101 telephone-event/8000
    a=sendrecv
    SIP/2.0 200 OK
    To: <sip:2221@10.146.7.2:5060>;tag=84cdf53da5a92854i0; intercom=true
    From: "Recpt Receptions"<sip:9050000000@10.146.7.2:5060;nf=v>;t ag=586ff962
    Call-ID: NTFhMGQ2YzQ3N2M0ZjI1ZWNhMWExM2Q3YmY4ZjMyOGU.
    CSeq: 4 INVITE
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-2a0eb676d5073411-1---d8754z-
    Contact: "Recpt Receptions" <sip:2221@10.146.7.135:5060>
    Server: Linksys/SPA962-5.1.15(aSC)
    Content-Length: 208
    Content-Type: application/sdp

    v=0
    o=- 4076014 4076017 IN IP4 10.146.7.135
    s=-
    c=IN IP4 10.146.7.135
    t=0 0
    m=audio 16388 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    ACK sip:2221@10.146.7.135:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-342ae95068286c0e-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:9050000000@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;tag=84cdf53da5a92854i0; intercom=true
    From: "Recpt Receptions"<sip:9050000000@10.146.7.2:5060;nf=v>;t ag=586ff962
    Call-ID: NTFhMGQ2YzQ3N2M0ZjI1ZWNhMWExM2Q3YmY4ZjMyOGU.
    CSeq: 4 ACK
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 0

    BYE sip:2221@10.146.7.135:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-991c342c7e265957-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:9050000000@10.146.7.2:5060>
    To: <sip:2221@10.146.7.2:5060>;tag=84cdf53da5a92854i0; intercom=true
    From: "Recpt Receptions"<sip:9050000000@10.146.7.2:5060;nf=v>;t ag=586ff962
    Call-ID: NTFhMGQ2YzQ3N2M0ZjI1ZWNhMWExM2Q3YmY4ZjMyOGU.
    CSeq: 5 BYE
    User-Agent: 3CXPhoneSystem 10.0.19534.0
    Content-Length: 0

    SIP/2.0 200 OK
    To: <sip:2221@10.146.7.2:5060>;tag=84cdf53da5a92854i0; intercom=true
    From: "Recpt Receptions"<sip:9050000000@10.146.7.2:5060;nf=v>;t ag=586ff962
    Call-ID: NTFhMGQ2YzQ3N2M0ZjI1ZWNhMWExM2Q3YmY4ZjMyOGU.
    CSeq: 5 BYE
    Via: SIP/2.0/UDP 10.146.7.2:5060;branch=z9hG4bK-d8754z-991c342c7e265957-1---d8754z-
    Server: Linksys/SPA962-5.1.15(aSC)
    Content-Length: 0

  6. #6
    Matthew's Avatar
    Matthew is offline Moderator
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    That was an inbound call, correct? What did you see as caller ID on this call? What should it have read?

    From: "Recpt Receptions"<sip:9050000000@10.146.7.2:5060;nf=v>;t ag=586ff962
    That from looks fine to me - I just need to know what you saw, as these logs look good.
    Matthew Hanson
    888Voip Systems Engineer
    www.888voip.com



  7. #7
    sanyaolu is offline Junior Member
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    That is what i see in wireshark too. But on the Phone we see "Recpt" That makes me confuse

  8. #8
    Matthew's Avatar
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    Are calls going to a Ring Group or Queue that is set to alter Caller ID?
    Matthew Hanson
    888Voip Systems Engineer
    www.888voip.com



  9. #9
    sanyaolu is offline Junior Member
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    The Calls come in from PRI => IVR => Ring-Group => Extension

    I did not setup anything to alter Caller ID. Where will i check that in 3CX

  10. #10
    Matthew's Avatar
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    If you route the call direct it should work. Ring Groups add Caller ID changes by default.

    Note in Settings=> General => Global Options that you can set the caller ID to be changed to be Prepended, or Appended, as well as whether to take Last name first or second.

    I bet you are prepending the group, which limits the length of displayed caller ID (15 Digits) so it cuts the last part.
    Matthew Hanson
    888Voip Systems Engineer
    www.888voip.com



  11. #11
    sanyaolu is offline Junior Member
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    Below is a snip of what i see in the Sangoma "GatewayDebug.log"



    pstn.in.callerName=ROB SMITH
    sip.out.from.displayName=ROB

    To: "9050000000" <sip:9050000000@10.146.7.2:5060>;tag=9078b030
    From: "ROB" <sip:4160000000@10.146.7.2:5066>;tag=pxip-callid-1310682926-482462-41-117ds-2cd6-6ec5a5c8

  12. #12
    Matthew's Avatar
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    Give this a try - from Netborder FAQ at Sangoma.com

    Caller name is not showing up.

    The caller name arrives in a FACILITY message and the gateway is configured to get the caller name from a DISPLAY IE in the SETUP message. To get the caller name right, you will need to change the path configuration of the gateway as follows:
    1. In the Pstn config tab of the Gateway Manager, choose "Call Control" and then "ISDN Configurations" in the panel on the left.
    2. Double-click on the Call control config name in the ISDN Configuration Grid (usually FAS1-T1)
    3. A Fas Configuration display menu will then appear, in the "Caller Name Location Method" scrolling menu, choose "IN-FACILITY-MSG"
    4. In the "Wair Facility Delay (ms)" menu enter 100
    5. Save the new configuration by clicking on the Save button.
    6. If you've got more than one Call Control Configuration, you will need to perform operations 3 to 5 for all your Call Control configurstions.
    7. Restart the gateway
    Matthew Hanson
    888Voip Systems Engineer
    www.888voip.com



 

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