Miguel Costa

That is correct - its posting to the application on the same server...via the following setting in pbxnsip: http://localhost:8162/pbxbroker/PBXCdrEndpoint

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

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

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

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?

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.

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

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