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Miguel Costa

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Posts posted by Miguel Costa

  1. Issues with CDR

    in CDR

    Posted

    That is correct - its posting to the application on the same server...via the following setting in pbxnsip:

     

    http://localhost:8162/pbxbroker/PBXCdrEndpoint

     

     

     

     

     

     

    Hi Miguel

     

    You don't explain how you have been exporting the inbound calls, however from the log i am going to assume they are sent to the application via xml as i can see a HTTP client connect after the call to HTTP client: Connect to 127.0.0.1:8162. Without knowing with accuracy which exporting method you are using i wont be able to point you in the right direction for troubleshooting.

  2. Issues with CDR

    in CDR

    Posted

    Need a bit of help here understanding the log.

     

    Here's the issue. We have a 3rd party reporting tool, that unfortunately stopped logging inbound calls on a per extension basis (we track agent call times, inbound is one of them). Here's an example test call i made to our trunk, to ext 118. Our vendor says they only report what the pbx sends to them - so basically something is broken at the pbx level or our sip provider??

     

    pbx ver 4.2.0.3981 (Win32)

     

     

    Here's a log of the call (intentionally hid my number for privacy reasons):

     

     

    >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

     

     

    [5] 2012/06/21 10:43:51: SIP Rx udp:64.183.104.146:56808:

    BYE sip:118@74.208.164.11:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 192.168.5.158:56808;branch=z9hG4bK56311c524dffbe90

    From: <sip:888976925X@192.168.101.100:5060;user=phone>;tag=94ff923159b02097

    To: <sip:480287499X@voicemail.pbxbox.com:5060;next6662pt01=-Next6662Pt01-i3l4psa6sctb6;user=phone>;tag=35002

    Supported: path

    Call-ID: 4b738cab@pbx

    CSeq: 3609 BYE

    User-Agent: Grandstream GXP2000 1.2.5.3

    Max-Forwards: 70

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

    Reason: SIP ;text="Onhook event"

    Content-Length: 0

     

    [5] 2012/06/21 10:43:51: SIP Tx udp:64.183.104.146:56808:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.158:56808;branch=z9hG4bK56311c524dffbe90;rport=56808;received=64.183.104.146

    From: <sip:888976925X@192.168.101.100:5060;user=phone>;tag=94ff923159b02097

    To: <sip:480287499X@voicemail.pbxbox.com:5060;next6662pt01=-Next6662Pt01-i3l4psa6sctb6;user=phone>;tag=35002

    Call-ID: 4b738cab@pbx

    CSeq: 3609 BYE

    Contact: <sip:118@74.208.164.11:5060;transport=udp>

    User-Agent: pbxnsip-PBX/4.2.0.3981

    Content-Length: 0

     

    [7] 2012/06/21 10:43:51: pcst13403006162216331037110@192.168.201.116: Media-aware pass-through mode

    [8] 2012/06/21 10:43:51: Hangup: Call 231 not found

    [5] 2012/06/21 10:43:51: SIP Tx udp:208.73.146.95:5060:

    BYE sip:480287499X@208.73.146.95:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 74.208.164.11:5060;branch=z9hG4bK-7a54b99c9571542ed9c348d7c080459a;rport

    From: <sip:888976925X@192.168.101.100:5060>;tag=f52d3141a3

    To: <sip:480287499X@192.168.101.116:5060;next6662pt01=-Next6662Pt01-i3l4psa6sctb6>;tag=SD77r9901-gK0a6461c0

    Call-ID: pcst13403006162216331037110@192.168.201.116

    CSeq: 27473 BYE

    Max-Forwards: 70

    Contact: <sip:442095058@74.208.164.11:5060;transport=udp>

    Content-Length: 0

     

    [8] 2012/06/21 10:43:51: HTTP client: Connect to 127.0.0.1:8162

    [8] 2012/06/21 10:43:51: Hangup: Call 231 not found

    [5] 2012/06/21 10:43:51: SIP Rx udp:208.73.146.95:5060:

    SIP/2.0 200 OK

    Via: SIP/2.0/UDP 74.208.164.11:5060;received=74.208.164.11;branch=z9hG4bK-7a54b99c9571542ed9c348d7c080459a;rport=5060

    From: <sip:888976925X@192.168.101.100:5060>;tag=f52d3141a3

    To: <sip:480287499X@192.168.101.116:5060;next6662pt01=-Next6662Pt01-i3l4psa6sctb6>;tag=SD77r9901-gK0a6461c0

    Call-ID: pcst13403006162216331037110@192.168.201.116

    CSeq: 27473 BYE

    Content-Length: 0

  3. Inbound / DID Issue

    in General Setup

    Posted

    PBX VER: 4.2.0.3981 (Win32)

     

    The instructions online seemed pretty straight fwd, but for the life of me, can't get this to work.

     

    We have added a DID number at our SIP provider, and its currently working fine and routing to the correct PBX trunk.

    However I would like to route this DID to a specific hunt group.

     

    >> HUNT GROUP

    >> ACCOUNT NUMBER(S): 892 323920035X

     

    When I call in, I reach the main AA and not the hunt group as expected. If I check the STATUS tab while on the call, I see:

     

    >> 07/28 3:53P Mike (+1480287499X) 892 connected Nextiva

     

    Visually its saying that it's sending the call to the proper hunt group 892, however why do I hear the AA instead (under 100)?? Here's the call log...

     

     

     

    >>>>>>>>>>>>>>>>>>>>>>>>>>>>

     

    Via: SIP/2.0/UDP 208.73.144.90:5060;branch=z9hG4bK3246c46c8424c8d763e22728d78271dd

    From: "Mike" <sip:+1480287499X@208.73.144.9X>;tag=3520880726-846450

    To: <sip:323920035X@208.73.144.9X:5060>;tag=f817fc8611

    Call-ID: 20534014-3520880726-846444@msc6.nextiva.com

    CSeq: 3 BYE

    Contact: <sip:538749811@74.208.79.73:506X;transport=udp>

    User-Agent: pbxnsip-PBX/4.2.0.3981

    Content-Length: 0

     

    <env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://soap.com/pbx"><env:Body><sns:CDR><PrimaryCallID>20534014-3520880726-846444@msc6.nextiva.com</PrimaryCallID><CallID>20534014-3520880726-846444@msc6.nextiva.com</CallID><From>"MIKE" <sip:+1480287499X@208.73.144.9X;user=phone></From><To><sip:323920035X@208.73.144.9X:5060;user=phone></To><Direction>I</Direction><Type>trunk</Type><AccountNumber>100@voicemail.PBXADDY.com</AccountNumber><RemoteParty>+1480287499X</RemoteParty><LocalParty></LocalParty><TrunkName>Nextiva</TrunkName><TrunkID>5</TrunkID><Domain>voicemail.PBXADDY.com</Domain><LocalTime>20110728152526</LocalTime><TimeStart>20110728222526</TimeStart><TimeConnected>20110728222527</TimeConnected><DurationHHMMSS>0:00:03</DurationHHMMSS><Duration>3</Duration><TimeEnd>20110728222530</TimeEnd><IPAdr>udp:208.73.144.9X:5060</IPAdr><Quality>VQSessionReport: CallTerm

    LocalMetrics:

    Timestamps:START=2011-07-28T22:25:27Z STOP=2011-07-28T22:25:30Z

    CallID:20534014-3520880726-846444@msc6.nextiva.com

    FromID:"Mike" <sip:+1480287499X@208.73.144.9X>;tag=3520880726-846450

    ToID:<sip:323920035X@208.73.144.9X:5060>;tag=f817fc8611

    SessionDesc:PT=0 PD=pcmu SR=8000 FD=20 FO=160 FPP=1 PPS=50 PLC=3

    LocalAddr:IP=74.208.79.7X PORT=52812 SSRC=0xeeebcf2b

    RemoteAddr:IP=208.73.144.86 PORT=20786 SSRC=0xddde9803

    x-UserAgent:pbxnsip-PBX/4.2.0.3981

    x-SIPterm:SDC=OK SDR=AN

  4. Constant Disconnects

    in General Setup

    Posted

    My users have been complaining of constant disconnects the last week or so. I'm also getting User disconnect emails to my admin account from the PBX. Here's the email content - any ideas what would cause this?

     

     

    >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>

     

     

    One side of the call between sip:442095058@sbc.voipdnsservers.com;user=phone and sip:9417804421@sbc.voipdnsservers.com;user=phone did not receive media for 2.5 s and the other side of the call disconnected the call. The address of the other side was 76.79.172.170 (User-Agent=Linksys/SPA941-5.1.8). You may use this email as hint for a potential problem. The SIP messages are attached.

     

    Rx: udp:76.79.172.170:31537 (756 bytes)

    INVITE sip:9417804421@pbxserver.com SIP/2.0

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-f6f91006

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 101 INVITE

    Max-Forwards: 70

    Contact: "149" <sip:149@192.168.5.121:5060>

    Expires: 240

    User-Agent: Linksys/SPA941-5.1.8

    Content-Length: 215

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

    Supported: replaces

    Content-Type: application/sdp

     

    v=0

    o=- 10341343 10341343 IN IP4 192.168.5.121

    s=-

    c=IN IP4 192.168.5.121

    t=0 0

    m=audio 16468 RTP/AVP 18 101

    a=rtpmap:18 G729a/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-15

    a=ptime:30

    a=sendrecv

    Tx: udp:76.79.172.170:31537 (328 bytes)

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-f6f91006;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 101 INVITE

    Content-Length: 0

     

    Tx: udp:76.79.172.170:31537 (542 bytes)

    SIP/2.0 401 Authentication Required

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-f6f91006;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 101 INVITE

    User-Agent: pbxnsip-PBX/4.0.1.3499

    WWW-Authenticate: Digest realm="pbxserver.com",nonce="6de691eb06c88b47d08550d9cfdc5303",domain="sip:9417804421@pbxserver.com",algorithm=MD5

    Content-Length: 0

     

    Rx: udp:76.79.172.170:31537 (419 bytes)

    ACK sip:9417804421@pbxserver.com SIP/2.0

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-f6f91006

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 101 ACK

    Max-Forwards: 70

    Contact: "149" <sip:149@192.168.5.121:5060>

    User-Agent: Linksys/SPA941-5.1.8

    Content-Length: 0

     

    Tx: udp:76.79.172.170:31537 (327 bytes)

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Content-Length: 0

     

    Tx: udp:76.79.172.170:31537 (859 bytes)

    SIP/2.0 183 Session Progress

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tx: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

    Tr: udp:76.79.172.170:31537 (845 bytes)

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.5.121:5060;branch=z9hG4bK-c3f8ef2;rport=31537;received=76.79.172.170

    From: "149" <sip:149@pbxserver.com>;tag=f2ec483d80c68b45o0

    To: <sip:9417804421@pbxserver.com>;tag=5b3fa43d56

    Call-ID: 40e1659d-d3e50d65@192.168.5.121

    CSeq: 102 INVITE

    Contact: <sip:149@74.208.77.157:5060>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

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

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/4.0.1.3499

    Content-Type: application/sdp

    Content-Length: 266

     

    v=0

    o=- 55038 55038 IN IP4 74.208.77.157

    s=-

    c=IN IP4 74.208.77.157

    t=0 0

    m=audio 55164 RTP/AVP 18 101

    a=rtpmap:18 g729/8000

    a=fmtp:18 annexb=no

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:30

    a=rtcp-xr:rcvr-rtt=all voip-metrics

    a=sendrecv

  5. Phones fail to register

    in Windows-related Topics

    Posted

    I had a Windows 2003 server provisioned by a hosting provider (Rackforce).

     

    The newest version of pbxnsip installed - and the interface is working fine. I currently manage a similar installation internally - but moving to a hosted scenario.

    Anyways long story short, the phones can't register. Turned off the firewall, I can access the admin page and such - but the phones for some reason fail to register. Any ideas?

  6. Branch Offices (Best setup?)

    in General Setup

    Posted

    We currently run an internally hosted pbxnsip server locally (Toronto) , but are planning to open a new sales office in Phoenix later this year.

    My concern is if I send all voice traffic to our Toronto office, I might need to eventually increase our bandwidth at this end. I would much rather have all outgoing calls routed directly to our SIP provider from Phoenix (about 90% of traffic is outgoing).

     

    Questions:

    1. Can I run another instance of pbxnsip from our branch office, and route all outbound calls via this server? I'm assuming this would mean I would use the outbound proxy settings (linksys phones) to point to the local server?

    2. Do I have to duplicate all extension settings? Or just configure a domain & trunk.

  7. Multiple Domain Setup

    in General Setup

    Posted

    in the trunks under "extension" put !([0-9]*)!\1!t!4164443333

    fyi .. the tel: may not even be necessary ... I just use the number ..

     

    I have run into this a few times lately with providers that do not require registration ..

     

    yori

    www.brandywinetech.com

     

    Yori,

     

    Thanks for that info. What do I place under the alias field for the auto-attendant?

     

    -M

  8. Multiple Domain Setup

    in General Setup

    Posted

    I understand the use of the tel: alias is required to properly route incoming calls to the appropriate auto-attendant.

     

    I have two separate domains > phones properly register on both sides.

     

    Trunks work fine for both domains, in other words, outgoing calls are successful. Both trunks for each domain is registering to the same provider. I use the remote-explicit-ID setting to manage the caller-id number for each domain. Again, outgoing is working fine.

     

    The issue I have is as soon as I configure a trunk in domain B, all incoming calls are directed to auto-attendent B, even if calling to domain A.

     

    Both trunks have been set with no extension setting. I use the tel: alias in the autoattendant for each domain, and use the ANI assigned by our provider (example: tel:4164443333). This does not work. I'm used to having our trunks register with a username that consists of our telephone number, but our new provider does not require a user or pw to register on their side.

     

    Questions:

    1. Would a tel: alias work in my case?

    2. If not, how the heck do I properly route the calls?

     

     

    Hopefully someone can point me in the right direction.

     

    -Miguel

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