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Caller ID problem

Discussion in 'Windows' started by Alan van der Vyver, Nov 23, 2009.

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  1. Alan van der Vyver

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    Hi!

    Our carrier requires that all calls originating from us have a "from" address that corresponds to one of our DID numbers (e.g. From: 5135952015@sigmatek.net).

    Our Sip server (SipX, not 3CX) is configured so that each account sends the correct caller-ID ("from" address) when using a hardware phone or the X-Lite software phone. However calls from the 3CX phone always have the extension or log-in Id as the from address. Changing the caller ID in the 3CX phones configuration changes the text part of the "from" field, but not the actual address. For example making the caller ID 5135952015 results in [From "5135952015" 2015@sigmatek.net]. This still results in outbound calls being refused by our carrier. Internal calls work fine.

    Does anyone know how to change the actual address used by the 3CX phone or how to make it use the server-configured address? It is otherwise quite a slick phone and I would like to use it.

    regards,
    Alan.
     
  2. rico06

    rico06 New Member

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    That was my question : I had outbound caller id in axtension but it doesnt change and my carrier accetp that

    why it doesnt work, is there anything else to change
     
  3. KerryG

    KerryG Active Member

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    You may need to adjust some of the advanced options for your outbound trunk, this should help.

    http://www.voipstore.com/how-to-set-3cx-caller-id/
     
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  4. Vali_3CX

    Vali_3CX Well-Known Member
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    Hi, Alan

    If I correctly understand (english is not my native language):
    - you cannot set 3CXPhone's extension field as 5135952015 and PBX extern IP field as sigmatek.net? In this case, supposing its caller ID field has been set as "Alan van der Vyver", 3CXPhone will identify itself as "Alan van der Vyver"<sip:5135952015@sigmatek.net>. I suppose 2015 it's (authentication) ID.

    Regards
    vali
     
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  5. Alan van der Vyver

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    Thanks for the information, but this relates to setting it in the 3CX server and I am not using a 3CX server. The problem I have is with the 3XC software phone.

    regards,
    Alan.
     
  6. Alan van der Vyver

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    Hi Vali,

    Thanks for the response.

    In this case the extension is 2015. It is also the log-in ID. The caller-ID that the carrier needs to see is 5135952015. That is also one of our direct inward dial (DID) numbers.

    The carrier needs something like: [From: "Alan van der Vyver" 5135952015@sigmatek.net].

    The 3CX software phone sends something like: [From: "Alan van der Vyver" 2015@sigmatek.net].

    Since 2015 is the internal number the carrier rejects the call.

    If I change the caller ID in the 3CX phone to: 5135952015, then it sends [From: "5135952015" 2015@sigmatek.net]. In this case the carrier still sees the call coming from 2015@sigmatek.net and rejects it.

    regards,
    Alan.
     
  7. Vali_3CX

    Vali_3CX Well-Known Member
    Staff Member 3CX Support

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    Hi, Alan

    All it's clear now, thanks. Well, 3CXPhone register itself in the form of caller_id<sip:extension@pbx>, so with the actual version there is no solution for your problem :roll: Obviously, it can be done in the future, since these parameters are available, only cannot be handled by user-interface; but, it's a "feature request" which should be discussed.

    However, 3CXPhone's basic rule is to be a standard SIP phone; from this point of view, I'm interested to see why there's a difference in behaviour compared with, let's say, X-Lite, which you said it works OK. In terms of 2015, 5135952015 and sigmatek.net, how did you set Xlite's "Display Name", "User name", "Authorization user name" and "Domain"? I only can see 5135952015, 2015, 2015, sigmatek.net, no difference compared with 3CXPhone. Or there is another setting in Xlite which allow you to get desired behavior?

    Thanks
    vali
     
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  8. Alan van der Vyver

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    Hi vali,

    I am also interested in why there is a difference. The only configuration difference I can find is that for the domain in X-Lite I just specify the domain (sigmatek.net) and it uses DNS SRV records to find the actual server, whereas on the 3CX phone I have to specify the actual host name or IP address. I can't imagine that this will make the difference though.

    I have syslog entries from the gateway for attempted calls from the 3CX phone and X-Lite and the difference in the "from" fields is visible in the first Invite, but there does not seem to be any other significant differences.

    I suspect I would have to take a debugging trace from the server when it is essentially idle to see what is actually happening in the interaction between the server and the software phones. I took a debugging trace from X-Lite and verified that it never knows anything about the 5135952015 caller-id that is configured in the server, but somehow that caller-ID shows up at the gateway when using X-Lite and it does not when using the 3CX phone.

    regards,
    Alan.
     
  9. Vali_3CX

    Vali_3CX Well-Known Member
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    Alan, how you configured Xlite's "Display name"? Empty or 5135952015?
    Also, 3CXPhone authentication is rejected at registration (REGISTER) time or at call (INVITE) time?
    Thanks
    vali
     
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  10. Alan van der Vyver

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    Hi!

    The display name is "Alan van der Vyver"
    The user name is "2015"
    The authorization name is "2015"
    The domain is "sigmatek.net"

    We should always remember here that I am not connecting to a 3CX server. I am connecting to a SipXecs server. However, it is a SIP server and X-Lite does work with it.

    I am not insisting that the 3CX phone work with the SipXecs server, but it would be nice if it did. It actually works very well with internal extensions. I only have a problem calling out.

    regards,
    Alan.
     
  11. Vali_3CX

    Vali_3CX Well-Known Member
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    Yes, I am aware of this, because we expect - as I said - 3CXPhone to act as a normal SIP phone, so it should work with any SIP server.
    I just edited my previous message, so I ask the second question again: 3CXphone is rejected at registration (REGISTER) time or at call (INVITE) time? I suspect you get a 403 error message
    Thanks
     
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  12. Alan van der Vyver

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    Hi vali,

    The 3CX phone registers OK with the server.

    You are correct that when the Invite is submitted to the gateway it returns 403 Forbidden.

    Here is part of the syslog from the gateway:

    2009-11-23 14:38:46 Local0.Notice 10.0.100.10 ( lgr_psbrdif)(1615646 ) pstn send --> PlaceCall: Trunk:0 BChannel:-1 ConnID:0 SrcPN=2015 SrcSN= DstPN=8860321 DstSN= SrcNT=0 SrcNP=0 SrcPres=0 SrcScrn=0 DstNT=0 DstNP=0 ServiceCap=M RdrctNum= RdNT=0 RdNP=0 RdPres=0 RdScrn=0 RdRsn=-1 Excl=-1 Display=Alan van der Vyver IE= UUIE=0,, CLIRReason:-1 OrigPN= OLI=-1 [Time: 15:38:49]
    2009-11-23 14:38:46 Local0.Notice 10.0.100.10 ( lgr_psbrdex)(1615647 ) pstn recv <-- CALL_PROCEEDING Trunk:0 Conn:0 BChannel:23 callhndl:0 Loc:-1 Des:-1 [Time: 15:38:49]
    2009-11-23 14:38:49 Local0.Notice 10.0.100.10 ( lgr_psbrdex)(1615648 ) pstn recv <-- CALL_DISCONNECTED Trunk:0 Conn:0 RetCause:104 NetCause:21 [Time: 15:38:52]
    2009-11-23 14:38:49 Local0.Notice 10.0.100.10 ( lgr_psbrdif)(1615649 ) UnNormal Disconnect cause:21#GWAPP_CALL_REJECTED Trunk:0 Conn:0 [Time: 15:38:52]
    2009-11-23 14:38:49 Local0.Notice 10.0.100.10 ( lgr_flow)(1615650 ) ---- Outgoing SIP Message to 10.0.100.11:5060 ---- [Time: 15:38:52]
    2009-11-23 14:38:49 Local0.Notice 10.0.100.10 SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 10.0.100.11;branch=z9hG4bK-sipXecs-6b2d256be6555eeef6a6cff7bdd6436b63da
    Via: SIP/2.0/UDP 10.0.100.11;branch=z9hG4bK-sipXecs-6b2ae4c2a409fd002eb71fbbdfb5056c4bcf%1af287e69eec34244c268757833f43aa
    Via: SIP/2.0/UDP 10.0.100.11;branch=z9hG4bK-sipXecs-6b25977d88033ce45a112b37447c96c116fd%a4109f98cf5205e7b23a06b8061ba2b7
    Via: SIP/2.0/UDP 10.0.0.131:64045;branch=z9hG4bK-d8754z-c41353182958870b-1---d8754z-;rport=64045
    From: "Alan van der Vyver"<sip:2015@10.0.100.11:5060>;tag=8217121f
    To: <sip:8860321@10.0.100.11:5060>;tag=1c1493853438
    Call-ID: MDkzNzIzOWQ4N2FjZDI1ZjNmY2RiN2NiZDFkYTliNjY.
    CSeq: 2 INVITE
    Record-Route: <sip:10.0.100.11:5060;lr;sipXecs-rs=%2Afrom%7EODIxNzEyMWY%60.400_authrules%2Aauth%7E%210c0f17b3fd6cf1d8e4368b15459544f0>
    Supported: em,timer,replaces,path,early-session,resource-priority
    Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
    Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.040.004
    Reason: Q.850 ;...

    The gateway is an AudioCodes Mediant 1000 at address 10.0.100.10 and the Sip server is at 10.0.100.11.

    The main difference you will find with X-Lite, apart from the fact that the call is not rejected, is that the from field is:

    From: "Alan van der Vyver"<sip:5135952015@sigmatek.net>;tag=e401b66a


    Where the 5135952015 is configured in the server as the caller-ID for extension 2015.

    regards,
    Alan.
     
  13. Vali_3CX

    Vali_3CX Well-Known Member
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    Hmm.. Alan, may you try to use, in 3CXPhone, instead of "sigmatek.net", its IP? You may get it by ping-ing sigmatek.net (I've got 66.161.148.181)
    I'm asking because, at one time, we had a 403 issue got on INVITE, also on a VOIP provider.
    Thanks
    vali

    P.S Also check this, maybe it's useful (I've searched for GWAPP_CALL_REJECTED)
    http://knowledge.3com.com/service/main.jsp;jsessionid=AD29ADA4B45DFC3A17E8D2F9B478AE24.selfservice1?t=solutionTab&ft=searchTab&ps=solutionPanels&locale=en_US&_dyncharset=UTF-8&curResURL=%2Fservice%2Fmain.jsp%3Bjsessionid%3DAD29ADA4B45DFC3A17E8D2F9B478AE24.selfservice1%3Ft%3DsearchTab%26locale%3Den_US%26_dyncharset%3DUTF-8%26SearchButton%3DFind%26focusTopic%3D%26topicName%3D%26useFocusTopic%3Dtrue%26searchstring%3DIncoming%252520anonymous%252520calls%252520are%252520being%252520disconnected%26sfield%3D%26dosearch%3Dtrue%26pn%3D87&solutionId=847&isSrch=Yes
     
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  14. Alan van der Vyver

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    Hi vali,

    I have tried using the host name and the IP address. It does not make any difference.

    Following the instructions in the link you sent, I can make outbound calls. The disadvantage of this is that we have more extensions that there are slots in the Audiocodes gateway table, so I can't put in a DID number for each person. If I am willing to have one caller ID for all outbound calls then we can easily set it up in the gateway. This may be a reasonable thing to do.

    What I was hoping for is picking up the caller ID from the SipX server and using that. That way each extension can have its caller ID equal to its DID number.

    I took a detailed trace on the SipX server last night to try to ascertain what the difference between the 3CX phone and X-Lite might be, but I don't know enough about the SIP protocol to be able to make good sense of it - particularly when it comes to interactions between the various services within the SipX server.

    The only difference I can see in the interaction is that an invite from the 3CX phone causes an extra nested call within the SipX server to another service (presumably the call resolver) and that results in an extra "via" leg in the invitation that is sent to the gateway. What is triggering that or why it should result in the loss of the server-defined caller-id, I don't know.

    I think that if this is going to be resolved at all, I shall have to take it up on the SipX server forum and see if anyone there can provide any useful information. If I hear anything useful, I will post it back here.

    Thanks for your help.

    regards,
    Alan.
     
  15. Vali_3CX

    Vali_3CX Well-Known Member
    Staff Member 3CX Support

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    Hi, Alan
    Alan, you're familiar with Wireshark? I'm asking because I would like to have two network captures, one for XLite and another for 3CXPhone, I guess only this could help to understand what's happening. If you can provide them, please PM me.

    Thanks (also) for your help 8)

    Regards
    vali
     
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  16. Alan van der Vyver

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    As a status update: After an off-line exchange with vali, it was determined to be a problem in the way the 3CX phone was registering with a SIP server that requires a domain specification.

    A patched, test version of the phone worked fine. I expect we will see this change in a future release of the phone.

    regards,
    Alan.
     
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