with a softphone yes, but with elastix no

Discussion in 'General' started by telecomtechnician, Aug 16, 2010.

  1. telecomtechnician

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    Hi elastix community

    There is a local VOIP provider in Venezuela (NETUNO), and I bought 4 VOIP sip lines from them. Iniatially they were configured in my elastix server (1.6)and after some ups and downs they worked fine for a while, but then they stopped registering to the provider. I thought it was a problem with the IPPBX, so I installed everything from scratch. For personal reasons, everynight I have to turn off the elastix server, until the next day. When I turned it on, sometimes the lines registers and the next day they do not. About 5 days ago I installed elastix 2.0 and configured the sip lines and they do not work.
    I decided to register the lines with the xlite softphone and !WELLA!, it registered fast with no problem.

    Definetely I have a mistake configuring the SIP trunk in elastix.

    I wait for your suggestions or ideas

    Thanks

    David Medina
     
  2. dicko

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    You might try a

    yum update

    unfortunately the Elastix 2.0 iso is largely dysfunctional and should not be used IMHO, an updated 2.0.1 iso was released a couple of days later when it was became patently apparent that they had maybe "jumped the gun" and totally ignored the concerns of many of the beta-testers. Maybe the yum update will get you functional, maybe you will need to wait a few "amounts of time" until the stable release is actually stable. I suggest you go back to 1.6 for production systems until the dust settles.

    regards

    dicko (one of those so ignored ;) )
     
  3. telecomtechnician

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    Thanks Dicko

    This failure is happening since elastix 1.6, anyway thanks for the tip.

    If my sip trunk is a "friend" type trunk, how can I configure that in elastix?, when I configure the trunk through the web interface (sip trunk) I have to fill outgoing settings and incoming settings and they are referred to "peer" and "user".

    Waiting for your comments

    David Medina
     
  4. dicko

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    basically

    peers can receive calls

    users can make calls

    friends can do both

    but to receive calls and if your provider does not provide IP based authentication, you need to "register" correctly with the provider so it knows where to send your calls to you, unfortunately that is not well defined and you should consult specifically with you provider as to what they require. If you have more than one DID you might need to either register individually your DID's as separate inbound trunks, or perhaps not, that is up to the provider.

    If you have host or IP based inbound calling, then you probably should not try to register at all, it will confuse Asterisk and perhaps the vendor.

    It is often easier with registrations based trunks, and quite acceptable, to provision several trunks with the same provider, one outbound (with no inbound parameters) , and a set of several inbounds (conversely without outbound parameters) to suit, call them SIPout and SIPin1, SIPin2 etc. as appropriate. This I find preferable, even with a single DID, as it isolates the specific function of each trunk (SIP trunks are NOT intrinsically bidirectional) , and is much easier to understand the logic thereof.
     
  5. telecomtechnician

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    Thanks again Dicko

    Based on your experience, answer me this:

    Why can I register in the first try with the xlite softphone with the sip trunk user account with no problems?

    I can dial in from any PSTN number to the DID assigned to this sip trunk and the xlite rings. I can dial out through the xlite to any PSTN number with no problems at all.

    So I know it is a configuration problem on my side (elastix server).

    This is the configuration parameters of one of my sip trunks

    21272xxxxx

    type=friend
    disallow=all
    allow=g729&alaw&ulaw&gsm
    host=216.22.81.49
    dtmfmode=rfc2833
    username=77118xxxxx
    fromuser=77118xxxxx
    secret=100xxx
    context=from-trunk
    canreinvite=no
    nat=yes
    qualify=yes
    insecure=very

    771180xxxxx:100xxx:771180xxxx@216.22.81.49/771180xxxx


    As I said, it is not something related to elastix 2.0, on elastix 1.6 the failure was an everyday basis, sometimes it registered well, but at night when I turned off the server and turn it on again the next day, there was no registration of the sip trunks. In the CLI> it says UNREACHABLE.

    Waiting for your comments

    David Medina
     
  6. dicko

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    Did you check with your vendor that the registration string should be


    7711xxxxxx:100xxx:7711xxxxxx@216.22.xx.xx/7711xxxxxx

    it looks dubious.

    and should likely be:

    user:secret@216.22.xx.xx/DID

    the user and DID might or might not be the same

    and please understand that if those are real values, you just let everybody in China use your trunks, at your cost, please edit your post immediately as necessary
     
  7. telecomtechnician

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    Hi Dicko

    Those were not real values, I changed some numbers. Anyway I just edited the post.

    So you say that the register line is very dubious, I said the same thing the first day I saw it.

    The local VOIP provider has an asterisk server on their side. What should I ask them so I can solve this issue?

    By the way, you did not answer me the question, why with a softphone yes and with elastix no?

    Once again thanks

    David Medina
     
  8. dicko

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    I didn't answer that because I can't, you didn't say what your registration string on the soft-phone was, either way that would be for your vendor to answer authoritatively, I can (and did) just point out that your asterisk registration string looks highly dubious as it has too many fields, maybe your soft-phone is cleverer than Asterisk, you, me and your vendor :)
     
  9. telecomtechnician

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    Xlite is clever definetely, lets make elastix too.

    Hi DICKO

    The technical guys of my local VOIP provider post me this screen debug and they say that my server is not sending to them, my username and password. How do I configure that in elastix?,

    I have done everything based on their configuration example, but there is something missing, what do you suggest?

    They configure sip trunk in the friend type.(not user or peer).

    Waiting for your comments

    (IP and username or changed)



    REGISTER sip:216.92.81.01 SIP/2.0

    Via: SIP/2.0/UDP 99.286.213.147:5060;branch=z9hG4bK56225488;rport

    Max-Forwards: 70

    From: <sip:XXXXXXX3@216.92.81.01>;tag=as5546d847

    To: <sip:XXXXXXX@216.92.81.01>

    Call-ID: 2513ee0350ab117f1e03ce5614746a85@192.168.0.3

    CSeq: 102 REGISTER

    User-Agent: Asterisk PBX 1.6.2.10

    Expires: 180

    Contact: <sip:XXXXXXX@99.286.213.147>

    Content-Length: 0





    SIP/2.0 401 Unauthorized

    Via: SIP/2.0/UDP 99.286.213.147:5060;branch=z9hG4bK56225488;received=99.286.213.147;rport=5060

    From: <sip:XXXXXXX@216.92.81.01>;tag=as5546d847

    To: <sip:XXXXXXX@216.92.81.01>;tag=as797ba6c4

    Call-ID: 2513ee0350ab117f1e03ce5614746a85@192.168.0.3

    CSeq: 102 REGISTER

    User-Agent: MIA_02

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

    Supported: replaces, timer

    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0729cb78"

    Content-Length: 0
     
  10. telecomtechnician

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    Xlite is clever definetely, lets make elastix too.

    Hi DICKO

    The technical guys of my local VOIP provider post me this screen debug and they say that my server is not sending to them, my username and password. How do I configure that in elastix?,

    I have done everything based on their configuration example, but there is something missing, what do you suggest?

    They configure sip trunk in the friend type.(not user or peer).

    Waiting for your comments

    (IP and username or changed)



    REGISTER sip:216.92.81.01 SIP/2.0

    Via: SIP/2.0/UDP 99.286.213.147:5060;branch=z9hG4bK56225488;rport

    Max-Forwards: 70

    From: <sip:XXXXXXX3@216.92.81.01>;tag=as5546d847

    To: <sip:XXXXXXX@216.92.81.01>

    Call-ID: 2513ee0350ab117f1e03ce5614746a85@192.168.0.3

    CSeq: 102 REGISTER

    User-Agent: Asterisk PBX 1.6.2.10

    Expires: 180

    Contact: <sip:XXXXXXX@99.286.213.147>

    Content-Length: 0





    SIP/2.0 401 Unauthorized

    Via: SIP/2.0/UDP 99.286.213.147:5060;branch=z9hG4bK56225488;received=99.286.213.147;rport=5060

    From: <sip:XXXXXXX@216.92.81.01>;tag=as5546d847

    To: <sip:XXXXXXX@216.92.81.01>;tag=as797ba6c4

    Call-ID: 2513ee0350ab117f1e03ce5614746a85@192.168.0.3

    CSeq: 102 REGISTER

    User-Agent: MIA_02

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

    Supported: replaces, timer

    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0729cb78"

    Content-Length: 0
     
  11. dingoland

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    As Dicko said to you, your registration string is not well formatted, you have too many characters.
    The correct format is : username:secret@voipdomain.tld/username or DID
    Some VOIPs accept only username:secret@voipdomain.tld.

    It is the case in France and working fine.

    For your trunk configuration, try enter all parameters in the peer details, let the user details empty and enter your registration string also in the registration field.

    Regards
    Greg
     

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