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kelvin

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Posts posted by kelvin

  2. V4 - Prepay Feature

    in Trunk Setup

    Posted

    Hi Support,

     

    The version 4 prepay feature sound good. May i know how to enable this feature? does it require additional license? Regard the credit control does it apply to domain level instead of account level. It mean entire domain sharing same credit amount. Thanks

     

    Regards,

     

    Kelvin

     

    Hamlet,

     

    Thanks for responding, any idea does the evaluation license come with prepay feature license? would like to try out before acquired.

     

    regards,

    Kelvin

  3. V4 - Prepay Feature

    in Trunk Setup

    Posted

    Hi Support,

     

    The version 4 prepay feature sound good. May i know how to enable this feature? does it require additional license? Regard the credit control does it apply to domain level instead of account level. It mean entire domain sharing same credit amount. Thanks

     

    Regards,

     

    Kelvin

  4. Multiple DDIs map to 1 SIP trunk

    in Microsoft OCS

    Posted

    Where are these DDIs assigned on the PBX? Also, can you please enable the SIP logging and send the trace to support@pbxnsip.com?

    (on how to enable the SIP tracing https://pbxnsipsupport.com/index.php?_m=kno...kbarticleid=58)

     

    hi support,

     

    we assigned in account number field by putting multiple value as below. i have emailed the log file. thanks

     

    +60327215890 60327215890 60327215891 60327215892 60327215893 60327215894 60327215895 60327215896 60327215897 60327215898 60327215899

  5. Multiple DDIs map to 1 SIP trunk

    in Microsoft OCS

    Posted

    So you want to keep the DID (DDI?) from the incoming call? There is a setting in the domain that instructs the PBX to leave the names unchanged. That might help.

     

    Support,

     

    the DID number unchanged, but pbxnsip sent multiple invite even we just called 1 DID number on. please looked below pbx log.

     

    we called DID +60327215891 but pbx sent all DID invite. we configure multiple DID in 1 extension.

     

    2009/09/02 16:03:49: Trunk IN-Cisco-94 (global) sends call to 60327215891 in domain donotdelete.com

    [9] 2009/09/02 16:03:49: Resolve 653334: url sip:+60327215890@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653334: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653334: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653335: url sip:+60327215899@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653335: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653335: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653336: url sip:+60327215891@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653336: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653336: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653337: url sip:+60327215892@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653337: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653337: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653338: url sip:+60327215893@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653338: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653338: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653339: url sip:+60327215894@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653339: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653339: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653340: url sip:+60327215895@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653340: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653340: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653341: url sip:+60327215896@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653341: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653341: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653342: url sip:+60327215897@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653342: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653342: tcp 202.168.66.202 5060

    [9] 2009/09/02 16:03:49: Resolve 653343: url sip:+60327215898@mycybhpv02.my.tria.de;transport=tcp

    [9] 2009/09/02 16:03:49: Resolve 653343: a tcp mycybhpv02.my.tria.de 5060

    [9] 2009/09/02 16:03:49: Resolve 653343: tcp 202.168.66.202 5060

    [5] 2009/09/02 16:03:49: Identify trunk (IP address match) 11

  6. Multiple DDIs map to 1 SIP trunk

    in Microsoft OCS

    Posted

    hi support,

     

    we have map multiple DDIs to a single extension and this particular extension we added manual registration for relay to OCS2007. routing path as below.

     

    DDIs --> SIP --> Pbxnsip --> SIP --> OCS 2007 R2

     

    When we call particular DDI, let said call DDI - 1111 but DDI - 2222, DDI - 3333 and so on... all DDI numbers will relay to OCS 2007 R2. OCS 2007 R2 receiving multiple SIP invite (we map 10DDIS, Pbxnsip will sent 10 SIP invite to OCS)

     

    please advise how can we configure to sent called DDI sip invite only. example: we call DDI - 2222 and pbxnsip only sent DDI - 2222 SIP invite only.

     

     

    regards,

    Kelvin

  8. Unknow RTP version 0

    in Microsoft OCS

    Posted

    hi guys,

     

    i am try to integrate pbxnsip with OCS 2007 R2. i have no problem making outgoing calls from OCS to pbxnsip but i receiving "unknow RTP version 0" return error in RTP packet from OCS when receiving incoming calls from pbxnsip. the phone will ring for few ring then drop call.

     

    i am using pbxnsip version 3.4.0.3201(win32) and OCS 2007 R2

     

    below is the wireshark packet captured.

     

    3 2009-08-25 20:07:35.838839 202.168.66.202 202.79.201.86 RTP Unknown RTP version 0

    Real-Time Transport Protocol

    00.. .... = Version: Old VAT Version (0)

     

    Any advice on this is truly appreciated. Thank you!

     

    Regards,

     

    Kelvin

  9. Incoming Caller id - Private no/hidden caller id

    in Trunk Setup

    Posted

    hi Support,

     

    The upgrade does not solve the problem, i still receiving caller id = Haniza () when incoming with not caller id and i notice the tel:xxx changed to +xxx after upgrade. what i suppose to do with +xxx? manual change to tel:xxx? thx

     

     

    hi support,

     

    what else we can do to resolve the caller id display issue. thx

     

    regards,

    kelvi

  10. Incoming Caller id - Private no/hidden caller id

    in Trunk Setup

    Posted

    Latest available Windows version is here - http://pbxnsip.com/protect/pbxctrl-3.1.1.3095.exe. Let us know if this version takes care of your problem.

    Also, make sure that you have backed up the working directory (a simple zip of pbx working directory is fine).

     

    hi Support,

     

    The upgrade does not solve the problem, i still receiving caller id = Haniza () when incoming with not caller id and i notice the tel:xxx changed to +xxx after upgrade. what i suppose to do with +xxx? manual change to tel:xxx? thx

  12. Incoming Caller id - Private no/hidden caller id

    in Trunk Setup

    Posted

    There was probably another problem with empty user names. If you can make the gateway use "anonymous@..." it would of course solve the problem. We'll try to incorporate a fix in the 3.1.1 version. If you want a preview, let me know what OS you have.

     

    hi Support,

     

    our pbxnsip is version 3.0.1.3023 on windows 2000 server. thx

     

     

    regards,

    kelvin

  14. Incoming Caller id - Private no/hidden caller id

    in Trunk Setup

    Posted

    In version 2.. caller id will display as private no. thx

     

    hi Support,

     

    In pbxnsip v3, when incoming call with no caller no, the caller id will change to weird name either domain name or extension name. please refer to follow log. thx

     

    INVITE sip:65014582@202.79.201.86:5080 SIP/2.0

    Via: SIP/2.0/UDP 202.79.201.83:5060

    From: <sip:202.79.201.83>;tag=170CFF74-1CC7

    To: <sip:65014582@202.79.201.86>

    Date: Wed, 19 Nov 2008 05:02:35 gmt

    Call-ID: 1D0F310D-B52E11DD-AFADC2C9-4499683E@202.79.201.83

    Supported: timer,100rel

    Min-SE: 1800

    Cisco-Guid: 487454909-3039695325-2947203785-1150904382

    User-Agent: Cisco-SIPGateway/IOS-12.x

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

    CSeq: 101 INVITE

    Max-Forwards: 6

    Remote-Party-ID: <sip:202.79.201.83>;party=calling;screen=no;privacy=off

    Timestamp: 1227070955

    Contact: <sip:202.79.201.83:5060>

     

    change to:

     

    INVITE sip:215@192.168.1.246 SIP/2.0

    Via: SIP/2.0/UDP 202.79.201.86:5080;branch=z9hG4bK-596d1b86ce9eb2f2eea733ce72ec8892;rport

    From: "Haniza" <sip:pbx.pgcomms.com;user=phone>;tag=4607

    To: "Kelvin Tee" <sip:215@pbx.pgcomms.com>

    Call-ID: 894b27a1@pbx

    CSeq: 27463 INVITE

    Max-Forwards: 70

    Contact: <sip:215@202.79.201.86:5080;transport=udp>

     

    my ip phone will display "haniza" as caller.

  15. Incoming Caller id - Private no/hidden caller id

    in Trunk Setup

    Posted

    hi Support,

     

    In pbxnsip v3, when incoming call with no caller no, the caller id will change to weird name either domain name or extension name. please refer to follow log. thx

     

    INVITE sip:65014582@202.79.201.86:5080 SIP/2.0

    Via: SIP/2.0/UDP 202.79.201.83:5060

    From: <sip:202.79.201.83>;tag=170CFF74-1CC7

    To: <sip:65014582@202.79.201.86>

    Date: Wed, 19 Nov 2008 05:02:35 gmt

    Call-ID: 1D0F310D-B52E11DD-AFADC2C9-4499683E@202.79.201.83

    Supported: timer,100rel

    Min-SE: 1800

    Cisco-Guid: 487454909-3039695325-2947203785-1150904382

    User-Agent: Cisco-SIPGateway/IOS-12.x

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

    CSeq: 101 INVITE

    Max-Forwards: 6

    Remote-Party-ID: <sip:202.79.201.83>;party=calling;screen=no;privacy=off

    Timestamp: 1227070955

    Contact: <sip:202.79.201.83:5060>

     

    change to:

     

    INVITE sip:215@192.168.1.246 SIP/2.0

    Via: SIP/2.0/UDP 202.79.201.86:5080;branch=z9hG4bK-596d1b86ce9eb2f2eea733ce72ec8892;rport

    From: "Haniza" <sip:pbx.pgcomms.com;user=phone>;tag=4607

    To: "Kelvin Tee" <sip:215@pbx.pgcomms.com>

    Call-ID: 894b27a1@pbx

    CSeq: 27463 INVITE

    Max-Forwards: 70

    Contact: <sip:215@202.79.201.86:5080;transport=udp>

     

    my ip phone will display "haniza" as caller.

  16. TEL ALIAS not working in version 3.0.0.2998

    in Extension Setup

    Posted

    .... try to use a ANI for the extension that wants to call the other name or a Prefix on the trunk. The PBX probably thinks the call comes from an extension, not from a trunk.

     

    by default the ANI = caller extension no right? i want to display Caller extension so that we can identify the calls. by the way what is this message mean? "[5] 2008/09/05 14:06:22: Received loopback request without tag"

  17. TEL ALIAS not working in version 3.0.0.2998

    in Extension Setup

    Posted

    Treat calls to other domains just like calls to other companies. As a trunk you can use the outbound proxy "127.0.0.1" - which loops the request back to itself. Make sure that this trunk is a "global" trunk, then you need only one.

     

    hi support,

     

    i using above method, and my snom return Authentication required with busy tone when try call interbranch extension. below is the log. thx

     

    SIP/2.0 183 Ringing

    Via: SIP/2.0/TLS 192.168.1.176:3101;branch=z9hG4bK-2v9nief449ot;rport=3101

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=dkm152o0i5

    To: <sip:365215@pbx.pgcomms.com.my;user=phone>;tag=539287bafe

    Call-ID: 3c3a1fd5d921-1d8euqo6pq07

    CSeq: 1 INVITE

    Contact: <sip:102@192.168.1.10:5081;transport=tls>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pgcomms-PBX/3.0.0.2998

    Require: 100rel

    RSeq: 1

    Content-Type: application/sdp

    Content-Length: 433

     

    v=0

    o=- 18443 18443 IN IP4 192.168.1.10

    s=-

    c=IN IP4 192.168.1.10

    t=0 0

    m=audio 54216 RTP/AVP 18 3 2 0 8 9 101

    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iIE4YWtoEmL5VNDncRdIeA2SKR2GKZly23Qc+k83

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:3 gsm/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

    a=sendrecv

    [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080:

    INVITE sip:65215@127.0.0.1:5080;user=phone SIP/2.0

    Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

    To: <sip:65215@127.0.0.1:5080;user=phone>

    Call-ID: 7b24e046@pbx

    CSeq: 18780 INVITE

    Max-Forwards: 70

    Contact: <sip:102@127.0.0.1:5080;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pgcomms-PBX/3.0.0.2998

    P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my>

    Content-Type: application/sdp

    Content-Length: 331

     

    v=0

    o=- 20980 20980 IN IP4 127.0.0.1

    s=-

    c=IN IP4 127.0.0.1

    t=0 0

    m=audio 58324 RTP/AVP 18 3 2 0 8 9 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:3 gsm/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:9 g722/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

    [5] 2008/09/04 17:26:40: Received loopback request without tag

    [7] 2008/09/04 17:26:40: UDP: Opening socket on port 57660

    [7] 2008/09/04 17:26:40: UDP: Opening socket on port 57661

    [5] 2008/09/04 17:26:40: Identify trunk (IP address/port match) 23

    [9] 2008/09/04 17:26:40: Resolve 17492: aaaa udp 127.0.0.1 5080

    [9] 2008/09/04 17:26:40: Resolve 17492: a udp 127.0.0.1 5080

    [9] 2008/09/04 17:26:40: Resolve 17492: udp 127.0.0.1 5080

    [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

    To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

    Call-ID: 7b24e046@pbx

    CSeq: 18780 INVITE

    Content-Length: 0

     

    [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

    To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

    Call-ID: 7b24e046@pbx

    CSeq: 18780 INVITE

    Content-Length: 0

     

    [9] 2008/09/04 17:26:40: Resolve 17493: aaaa udp 127.0.0.1 5080

    [9] 2008/09/04 17:26:40: Resolve 17493: a udp 127.0.0.1 5080

    [9] 2008/09/04 17:26:40: Resolve 17493: udp 127.0.0.1 5080

    [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

    To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

    Call-ID: 7b24e046@pbx

    CSeq: 18780 INVITE

    User-Agent: pgcomms-PBX/3.0.0.2998

    WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5

    Content-Length: 0

     

    [7] 2008/09/04 17:26:40: SIP Rx udp:127.0.0.1:5080:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport=5080

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

    To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

    Call-ID: 7b24e046@pbx

    CSeq: 18780 INVITE

    User-Agent: pgcomms-PBX/3.0.0.2998

    WWW-Authenticate: Digest realm="pbx.pgcomms.com.my",nonce="4cf3b7a72fbee5868e7e410c0e4ee2e4",domain="sip:65215@127.0.0.1:5080;user=phone",algorithm=MD5

    Content-Length: 0

     

    [7] 2008/09/04 17:26:40: Call 7b24e046@pbx#15046: Clear last INVITE

    [9] 2008/09/04 17:26:40: Resolve 17494: url sip:127.0.0.1:5080

    [9] 2008/09/04 17:26:40: Resolve 17494: udp 127.0.0.1 5080

    [7] 2008/09/04 17:26:40: SIP Tx udp:127.0.0.1:5080:

    ACK sip:65215@127.0.0.1:5080;user=phone SIP/2.0

    Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-adf90a1753c88364b9213f3cfa2f4e48;rport

    From: "Normala" <sip:102@pbx.pgcomms.com.my>;tag=15046

    To: <sip:65215@127.0.0.1:5080;user=phone>;tag=c054a4dbeb

    Call-ID: 7b24e046@pbx

    CSeq: 18780 ACK

    Max-Forwards: 70

    Contact: <sip:102@127.0.0.1:5080;transport=udp>

    P-Asserted-Identity: "Normala" <sip:102@pbx.pgcomms.com.my>

    Content-Length: 0

  19. TEL ALIAS not working in version 3.0.0.2998

    in Extension Setup

    Posted

    We had to change a couple of things with the alias. The calls must now be routed through a trunk to the other domain. The old way is not suitable for server farms where you have no idea on which server a domain (or tel:-alias) is physically located. As we catch up with the documentation we'll elaborate that on the Wiki in further detail.

     

    for this case, what alternate way i can use to route inter branch calls. thx

     

    regards,

    kelvin

  21. Codec Code

    in General Setup

    Posted

    G.723 is not supported by pbxnsip. This codec is not worth it in a PBX environment:

    • Bad audio quality (well, what can you expect from 5.3 kbit/s?)
    • High CPU load (big problem for central network equipment that does support MoH and call barge in, recording)
    • Show-stopper license terms (lots of patents, unclear situation who will take us to court us if we support it)

    And don't forget that 5.3 kbit/s does not mean that a RTP session takes 5.3 kbit/s. The RTP packet overhead is somewhere in the 12-24 kbit/s area, depending on the packet size. Yepp, that's 3 times higher than the compressed audio information itself. Making G.723 even more pointless.

     

    ok. thanks

     

    regards,

    kelvin

  22. DTMF - SIP info

    in Trunk Setup

    Posted

    The support is only for receiving DTMF INFO, no transcoding between signalling layer and media layer. This is just because we don't like to make our live miserable. Think about someone sending RFC2833 (=RFC4833) DTMF tones and then while the tone is playing back someone sends a INFO as well.

     

    Plus the support for RFC2833 tones is pretty good these days.

     

    ok. thx

     

    regards,

    kelvin

  23. DTMF - SIP info

    in Trunk Setup

    Posted

    hi support,

     

    may i know pbxnsip support dtmf sip info relay? if yes, how to change it on trunk? we experience dtmf issue on quintum DX2030 with dtmf sip info relay. when call connect to IVR system, dtmf does not recognize by ivr system. if change to h245 outband then no issue but due to environment setup we cant use h245 outband. thx

     

    regards,

    kelvin

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