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SIP REGISTER bug**UNSUPPORTED PROVIDER

Discussion in '3CX Phone System - General' started by jitdor, Jul 10, 2010.

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  1. jitdor

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    I was given the following credentials for my SIP account which I plan to use with my 3CX Phone System:

    Line number: XXX
    Auth ID: YYY
    Auth pw: ZZZ

    For the line to work, my provider requires the "Contact", "From", and "To" SIP headers to use XXX as the user part. YYY is only used during authentication digest.

    However, 3CX sends "YYY" for all the "Contact", "From" and "To" fields during SIP REGISTRATION. And I don't seem to be able to change that at all.

    I am currently using v9 RC1 (9.13373).

    SIP INVITE fields are changeable under the Outgoing tab, which I did. I sniffed the SIP packets during an outgoing call which seems to be correct. However, my provider does not allow making a call without a successful SIP registration -- which means I am stuck until I can customize the fields for registration.

    Please help!
     
  2. LeonidasG

    LeonidasG Support Team
    Staff Member 3CX Support

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    Re: SIP REGISTER bug

    Hi,

    What's the name of your VoIP Provider?
    I take it you already tried creating a Generic VoIP Provider / Generic VoIP Trunk.

    If your VoIP Provider cant' register at all, i'd check your stun settings in the Phonesystem or whether you can even reach the internet from the machine 3CX is installed on.
     
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  3. jitdor

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    Re: SIP REGISTER bug

    Thanks LeonidasG for the quick reply.

    STUN settings on the 3CX server are working fine. I have another two trucks that are functioning normally, for both incoming and outgoing.

    Like I've mentioned in my earlier post, the problem is with 3CX not passing the correct parameters during REGISTER.

    My provider is starhub from Singapore. It uses different fields for username and auth id.

    Username is my line number (let it be XXX). But the Auth ID is a different number (YYY).

    During successful registrations, as captured by Winshark, the following information is sent in the SIP message header:
    Contact: <XXX@MyInternalIP:5060>
    To: <XXX@GWHost:5060>
    From: <XXX@GWHost:5060>
    Authentication: Digest username="YYY"...

    However, for 3CX, it always send:
    Contact: <YYY@MyInternalIP:5060>
    To: <YYY@GWHost:5060>
    From: <YYY@GWHost:5060>
    Authentication: Digest username="YYY"...
    (Note the YYY instead of XXX fields for Contact/To/From)

    resulting in a "404 User not found" error.

    In short, my provider requires SIP registrations to use my line number for the Contact/To/From fields, and the authID for authentication digest. Any other combinations would result in 403 or 404 errors.

    There seems to be no way for me to change this REGISTER behavior in 3CX...
     
  4. nb

    nb Support Team
    Staff Member 3CX Support

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    Re: SIP REGISTER bug

    All providers we support use only Authentication ID and Authentication Password.
    It seems that this provider uses 3 way authentication. At the moment it is not supported.

    Unfortunately there is nothing you can do to tweak this behavior.

    As you said the outbound parameters are there to manipulate the invite. The registration process is hardcoded.

    Can your provider offer you a trunk instead? No authentication required but instead it is bound to your IP Address.
     
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  5. jitdor

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    Re: SIP REGISTER bug

    Thanks nickybrg for the definitive clarification.

    The SIP account is given to us (home broadband users) free, via a Motorola IAD. Trunking is not possible as we get dynamic IP addresses via DHCP -- and the service is supposed to be portable -- users can bring the IAD or softphone around the world and make calls. My goal is to integrate this SIP line into a PBX. 3CX is the first choice as the free version allows multiple VoIP lines and I am also running a Windows server. Running a virtualized CentOS-based PBX seems to create server-time accuracy problems.

    Right now I'm using the account on a Linksys SPA-942 that registers correctly and works fine, but the other IP phones in the house (including two CP-7961G and a D2Tech adapter which are connected to 3CX) can't make use of that line -- which is quite an inconvenience as that happens to be the most-frequently-used line.

    So if the hardcoded registration process is not going to be changed anytime soon, or that there's no workaround, I may need to try something else.
     
  6. carolinainnovative

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    Re: SIP REGISTER bug

    stupid question nicky - with the contact, to and from fields hardcoded using outbound params, and the authid and password set properly... why wouldn't the registration use the outbound params? or at least be able to configure them in some form or fashion similar to the outbound params. That seems like a decent sized OVERSIGHT on 3cx's part...
     
  7. nb

    nb Support Team
    Staff Member 3CX Support

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    Re: SIP REGISTER bug**Unsupported Voip Provider

    This is how it is at the moment. There is nothing to do. Maybe when the demand for this increases we will consider creating the ability to create registrations like this. All the supported providers that we work with do not use this behavior. In such wide standards of operation we cannot support everything.

    Yes - on the other hand the provider can put the user name in the request uri and the auth id in the to from and contact fields.
    RFC says it this way. Request uri and To should be different from each other. And To, from and optional contact should have same user part. As you can see this is also valid. Also the most used.
     
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  8. ivan

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    I have the same problem. The provider settings are:

    Line number: XXX
    Auth ID: XXX@provider_host
    Auth pw: ZZZ

    Authentication is not possible, because Auth ID is copied to fields: To, From and Contact in register message. We are making another ippbx based on a SIP stack where the register phase is going fine, because the auth ID and user ID (line number) are treated separately.
    It would be nice if 3CX could introduce new tag for authentication parameters simmilar to Inbound and Outbound ones.
     
  9. nb

    nb Support Team
    Staff Member 3CX Support

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    We are going to implement this in version 10.

    Can you tell me the voip prvider you are using please? I am gathering a list of voip providers that support 3 way authentication.
     
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  10. jitdor

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    That's great news nickybrg! I'd be more than willing to pilot/beta test and feedback.
    My VoIP provider is StarHub from Singapore.
     
  11. nb

    nb Support Team
    Staff Member 3CX Support

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    Perfect!!
    Then we will keep in touch - Around January we are going to have something we can start testing.

    Send me an email at nb@3cx.com
     
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  12. nzvoipguy

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    Hi Nicky,

    Do you currently have a scheduled release date for version 10? I work for a VoIP provider in New Zealand and our standard practice is to use 3 way auth as well so I would be interested in testing this.

    Thanks
     
  13. nb

    nb Support Team
    Staff Member 3CX Support

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    Then I will gladly contact you in this case. Around January 2011 we will start having the first alphas.
     
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  14. JJtenner

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    Hi Nicky, I also have the same issue with my voip provider in new zealand with 3 way id so 3cx is unusable unless I Sip trunk which I dont have the demand for. Any chance you are keen for another beta tester in NZ as I love your software but lose all but interanal functionality due to the non support of 3 way, . Thanks
     
  15. nb

    nb Support Team
    Staff Member 3CX Support

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    Hi There - no probs. Send me an email nb@3cx.com and I will send you a build with 3Way authentication support.
     
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  16. gold_god

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    Hi Nicky,

    I have extacly the same issues for years with 3CX. This will be a great feature.

    Currently I cannot configure 3CX to have an AuthID (u10419615) seperated from UserID (010419615).
    I'll be happy to test a build with 3Way authentication support with my provider and give you the feedback you need if I could.

    My provider is Belgacom (biggest telecom operator in Belgium, land-lines/mobile/internet/TV/international carrier). I cannot connect to my I-Talk line because of that.

    Cheers,

    Mar 14 08:19:40.031 : Recv 562 bytes from 85.158.211.211:17056 :
    REGISTER sip:voip.belgacom.be SIP/2.0
    Via: SIP/2.0/UDP 127.0.0.1:17056;branch=z9hG4bK-d8754z-c6d79473d3fe738a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:010419615@127.0.0.1:17056;rinstance=627715c0dac108e9>
    To: "I-Talk"<sip:010419615@voip.belgacom.be>
    From: "I-Talk"<sip:010419615@voip.belgacom.be>;tag=c3af96ec
    Call-ID: MjRlYTg0ZmU0YTc0NjZjY2FiYmJmNDhjNmNkZDc1NTc.
    CSeq: 1 REGISTER
    Expires: 3600
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    User-Agent: X-Lite 4 release 4.0 stamp 58832
    Content-Length: 0

    Mar 14 08:19:40.171 : Recv 523 bytes from 81.240.251.38:5060 :
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 127.0.0.1:17056;received=85.158.211.211;branch=z9hG4bK-d8754z-c6d79473d3fe738a-1---d8754z-;rport=2906
    To: "I-Talk"<sip:010419615@voip.belgacom.be>;tag=SD1mkn299-5e31ba2a-0014-002f-0000-0000
    From: "I-Talk"<sip:010419615@voip.belgacom.be>;tag=c3af96ec
    Call-ID: MjRlYTg0ZmU0YTc0NjZjY2FiYmJmNDhjNmNkZDc1NTc.
    CSeq: 1 REGISTER
    Server: SSW/0.0.0
    WWW-Authenticate: Digest realm="voip.belgacom.be",nonce="4d7dc0efdb6a7291fe26df40b18d1372c6a0afed",algorithm=MD5
    Content-Length: 0

    Mar 14 08:19:40.171 : Recv 765 bytes from 85.158.211.211:17056 :
    REGISTER sip:voip.belgacom.be SIP/2.0
    Via: SIP/2.0/UDP 127.0.0.1:17056;branch=z9hG4bK-d8754z-9fa2715b3405c124-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:010419615@127.0.0.1:17056;rinstance=627715c0dac108e9>
    To: "I-Talk"<sip:010419615@voip.belgacom.be>
    From: "I-Talk"<sip:010419615@voip.belgacom.be>;tag=c3af96ec
    Call-ID: MjRlYTg0ZmU0YTc0NjZjY2FiYmJmNDhjNmNkZDc1NTc.
    CSeq: 2 REGISTER
    Expires: 3600
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    User-Agent: X-Lite 4 release 4.0 stamp 58832
    Authorization: Digest username="u10419615",realm="voip.belgacom.be",nonce="hash_deleted",uri="sip:voip.belgacom.be",response="hash_deleted",algorithm=MD5
    Content-Length: 0
     
  17. gold_god

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    Thank you very much Nicky, it works flawlessly with in 3CXPhoneSystem 10.0.17590.1350

    I'm just waiting for v10 to be released now ;-)

    Cheers.
     
  18. nb

    nb Support Team
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    Thanks for the update
     
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  19. xswoosh

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    Hi,

    I am configuring 3CX to connect to my SIP account with my provider Starhub.

    I'm seeking help to determine the right information to enter into 3CX.

    Similar to the earlier message in this thread, I have the following information that works with my existing SIP Softphone that is configured to my SIP Provider:

    Line number: XXX
    Auth ID: YYY
    Auth pw: ZZZ


    During successful registrations, as captured by Winshark, the following information is sent in the SIP message header:
    Contact: <XXX@MyInternalIP:5060>
    To: <XXX@GWHost:5060>
    From: <XXX@GWHost:5060>
    Authentication: Digest username="YYY"...

    However, for 3CX, it always sends:
    Contact: <YYY@MyInternalIP:5060>
    To: <YYY@GWHost:5060>
    From: <YYY@GWHost:5060>
    Authentication: Digest username="YYY"...
    (Note the YYY instead of XXX fields for Contact/To/From)

    From the earlier message on the thread, I understand the new 3CX V10 3-way authentication ID fixes the following problem. However, I get an error in the Server Activity Log that says - [CM102001]: Authentication failed for SipReq: REGISTER 192.168.1.102 tid=d25cec084d178c57 cseq=REGISTER contact=DialCode@127.0.0.1:5488 / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings.

    1) Could someone advise me if the following is the correct information to enter into the 3CX VOIP Service Provider Account Details?
    External Number: XXX
    Authentication ID: XXX
    Authentication Password: ZZZ
    3-way Authentication ID: YYY

    2) Could you also let me know what is the best way to test if the VOIP Service Provider configuration is correct. I currently stop the Phonesystem Service and start it again. And check the Server Activity log for any success. Is this the way to check if the configuration works?

    Thanks.
     
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