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Det

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  1. Okay, we did a simulation without the a= header on our phone here and maybe the following version fixes that problem:

     

    http://www.pbxnsip.com/download/pbxctrl-2.1.0.2117.exe

     

    We found something that would explain why the 0.0.0.0 method would not work. Please verify.

     

    This version did the trick!! Yeah... my MoH is working again with the 9133i phones!!

     

    Thanks alot for the fast help adapting to the old style!

     

    Detlef

  2. Put the .2116 in this morning but there seems to be a bug - its still not working! It keeps asking for "Authentication Required" and reports "Password does not match". Below the logfile with a call from a X-Lite softphone to an Aastra 9133i when the Aastra was trying to put the X-Lite on hold:

     

     

     

    [9] 2007/10/15 07:43:51: SIP Rx udp:192.168.104.222:5060:

    INVITE sip:101@192.168.104.220:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK24eefb4a0

    Max-Forwards: 70

    Content-Length: 265

    To: "X-Lite Softphone" <sip:150@localhost>;tag=27498

    From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d

    Call-ID: 1e63199b@pbx

    CSeq: 953138503 INVITE

    Supported: timer

    Allow-Events: talk,hold,conference

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

    Content-Type: application/sdp

    Contact: D.Schade <sip:101@192.168.104.222>

    Supported: replaces

    User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

     

    v=0

    o=MxSIP 0 369883472 IN IP4 192.168.104.222

    s=SIP Call

    c=IN IP4 0.0.0.0

    t=0 0

    m=audio 3000 RTP/AVP 0 8 18 2 101

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:18 G729/8000

    a=rtpmap:2 G726-32/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

     

    [9] 2007/10/15 07:43:51: Resolve destination 190: a udp 192.168.104.222 5060

    [9] 2007/10/15 07:43:51: Resolve destination 190: udp 192.168.104.222 5060

    [9] 2007/10/15 07:43:51: SIP Tx udp:192.168.104.222:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK24eefb4a0

    From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d

    To: "X-Lite Softphone" <sip:150@localhost>;tag=27498

    Call-ID: 1e63199b@pbx

    CSeq: 953138503 INVITE

    Contact: <sip:101@192.168.104.220:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.0.2116

    Content-Type: application/sdp

    Content-Length: 273

     

    v=0

    o=- 6791 6791 IN IP4 192.168.104.220

    s=-

    c=IN IP4 192.168.104.220

    t=0 0

    m=audio 54610 RTP/AVP 0 8 18 2 101

    a=rtpmap:0 pcmu/8000

    a=rtpmap:8 pcma/8000

    a=rtpmap:18 g729/8000

    a=rtpmap:2 g726-32/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendrecv

     

    [9] 2007/10/15 07:43:52: SIP Rx udp:192.168.104.222:5060:

    ACK sip:101@192.168.104.220:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 192.168.104.222;branch=z9hG4bK00cb18415

    Max-Forwards: 70

    Content-Length: 0

    To: "X-Lite Softphone" <sip:150@localhost>;tag=27498

    From: "Aastra 9133i" <sip:101@localhost>;tag=48467520087b91d

    Call-ID: 1e63199b@pbx

    CSeq: 953138503 ACK

    Contact: D.Schade <sip:101@192.168.104.222>

    User-Agent: Aastra 9133i/1.4.2.1081 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45

     

     

    [9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055:

    SUBSCRIBE sip:192.168.104.220 SIP/2.0

    Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport

    From: <sip:101@ims-va.com>;tag=2766

    To: <sip:101@ims-va.com>

    Call-ID: zycdrj93q2jke09ah8rj

    CSeq: 10551 SUBSCRIBE

    Max-Forwards: 70

    Contact: <sip:101@0.0.0.0:1055>

    Event: x-tapi

    Accept: application/x-tapi

    Expires: 3600

    Content-Length: 0

     

     

    [9] 2007/10/15 07:43:57: Last message repeated 2 times

    [9] 2007/10/15 07:43:57: Message repetition, packet dropped

    [9] 2007/10/15 07:43:57: Resolve destination 192: udp 192.168.104.129 1055

    [9] 2007/10/15 07:43:57: SIP Tx udp:192.168.104.129:1055:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129

    From: <sip:101@ims-va.com>;tag=2766

    To: <sip:101@ims-va.com>;tag=0b4a71dab2

    Call-ID: zycdrj93q2jke09ah8rj

    CSeq: 10551 SUBSCRIBE

    User-Agent: pbxnsip-PBX/2.1.0.2116

    WWW-Authenticate: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",domain="sip:192.168.104.220",algorithm=MD5

    Content-Length: 0

     

     

    [9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055:

    SUBSCRIBE sip:192.168.104.220 SIP/2.0

    Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport

    From: <sip:101@ims-va.com>;tag=2766

    To: <sip:101@ims-va.com>

    Call-ID: zycdrj93q2jke09ah8rj

    CSeq: 10552 SUBSCRIBE

    Max-Forwards: 70

    Contact: <sip:101@0.0.0.0:1055>

    Event: x-tapi

    Accept: application/x-tapi

    Authorization: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",response="067ab894d4b5db7e1cc860d426c2109d",username="101",uri="sip:192.168.104.220",algorithm=MD5

    Expires: 3600

    Content-Length: 0

     

     

    [9] 2007/10/15 07:43:57: Resolve destination 193: udp 192.168.104.129 1055

    [9] 2007/10/15 07:43:57: SIP Tx udp:192.168.104.129:1055:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129

    From: <sip:101@ims-va.com>;tag=2766

    To: <sip:101@ims-va.com>;tag=0b4a71dab2

    Call-ID: zycdrj93q2jke09ah8rj

    CSeq: 10552 SUBSCRIBE

    User-Agent: pbxnsip-PBX/2.1.0.2116

    Warning: 399 ims-va.com Password does not match

    Content-Length: 0

     

     

    [9] 2007/10/15 07:43:57: SIP Rx udp:192.168.104.129:1055:

    SUBSCRIBE sip:192.168.104.220 SIP/2.0

    Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport

    From: <sip:101@ims-va.com>;tag=2766

    To: <sip:101@ims-va.com>

    Call-ID: zycdrj93q2jke09ah8rj

    CSeq: 10552 SUBSCRIBE

    Max-Forwards: 70

    Contact: <sip:101@0.0.0.0:1055>

    Event: x-tapi

    Accept: application/x-tapi

    Authorization: Digest realm="ims-va.com",nonce="4f52fd692f715e2d6737406f9e800f2c",response="067ab894d4b5db7e1cc860d426c2109d",username="101",uri="sip:192.168.104.220",algorithm=MD5

    Expires: 3600

    Content-Length: 0

     

     

    [9] 2007/10/15 07:43:57: SIP Tm udp:192.168.104.129:1055:

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP 0.0.0.0:1055;branch=z9hG4bK-t6krrzox8ju4sydosl7x;rport=1055;received=192.168.104.129

    From: <sip:101@ims-va.com>;tag=2766

    To: <sip:101@ims-va.com>;tag=0b4a71dab2

    Call-ID: zycdrj93q2jke09ah8rj

    CSeq: 10552 SUBSCRIBE

    User-Agent: pbxnsip-PBX/2.1.0.2116

    Warning: 399 ims-va.com Password does not match

    Content-Length: 0

     

     

    [9] 2007/10/15 07:43:57: Message repetition, packet dropped

  3. Hello Specialists,

     

    I am looking to implement a VoIP PBX system for our world wide locations and I am wondering if and how pbxnsip would be used/recommended to accomplish this.

     

    General data:

     

    - 4 locations (3 US, 1 Mexico) that should use pbxnsip

    - 1 Headquater in Germany with Innovaphone IP305 as gateway

    (works with the pbxnsip demo already nicely)

    - each US/Mex location has about 4-8 PSTN lines and 20 extensions

    - all locations are connected via VPNs with each other

    - each location needs to maintain local PSTN numbers

     

    Questions:

     

    Anyone using a pbxnsip setup like that yet?

    Is it the best solution to place a pbxnsip at each location?

    How do multiple pbxnsip exchange data with each other (easy extension dialing)?

    How stable would be one central pbxnsip work?

     

    Thanks in advance

    Det

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