zazi

Hi, I don't know really what do you mean with "inband DTMF" (it is just my first VUI during my studies). If it is related to the usage of DTMF, then my experience is that DTMF works for inbound calls. For outbound calls it is currently not implemented (so can't test it yet), but we've planned to link the main part of the workflow of our inbound VUI to our outbound VUI. Generally, our VUI is designed for natural language queries/ communication. So DTMF is just a helping component in the case that the user isn't able to do a natural language query that matches. Cheers zazi

Hello again, now we got also the outbound call running. Here is a short description how to it (please ask for a detail one, if necessary): 1. set the proxy at the makeCall activity to your SIP peer of your pbx(nsip) 2. create an account with the specified name of your calling party (from the makeCall activity) 3. route the outbound call to your outbound proxy (I used here my sipgate account) - it runs here as "Outbound Proxy" - maybe also important to set up the port of the outbound server Cheers zazi PS: I'm still running version 3.0.1.3023 (because I've read about the problems with P-Asserted-Identify)

Hi Jan, through the Sipgate VoIP provider we connected our system to a real telephone number so you can use every kind of telephone or landline. That was a prior design goal of our application. Of course, you can also use a softphone client (in ways: 1. you call directly the delegated telephone number 2. you register the softphone client directly at the ip-pbx. The outgoing call should go over the Message Queuing system, which is related to the hosted speech application. And then know way over the system: the ip-pbx delegated it to the sipgate phone number and the initiate the real call to a real number. Unfortunatelly, this is currently not implemented in our system. We are just sending SMS outside, but therefore we use directly the Webservice from Sipgate. Cheers zazi PS: sorry for the late reply, I've disabled the email notification

Oh, I think dsl modem tries do resolved them and when it can not resolve the hostname, it routes it to a further machine outside. The issue is that without a phone call the registration refreshes all the time successfully, it is still only after a phone call. However, I've tried it at my university and it works - so it is only a local problem of our combination/ configuration at home (because if every thing of our architecture fits - the combination of PBXNSIP + MS Speech Server etc. - we get a Windows Server 2003 at the university, where we can host our applications). Thanks a lot for all your help. Cheers zazi

Thanks a lot for the information about STUN. Nevertheless I'm running my system (as I maybe decribed in my post above) with a keep-alive time of 30 seconds. At the beginning it works really good and renew the registration after every 30 seconds. When I dial my sipgate number from my hardware phone to connect to my MS Speech Server application (yeah, the routing works now fine), the registration get lost after this call and only a reboot of my system with in general a new IP from my router (I used DHCP locally) will help. Anyway, that can't be the solution. When I have a look at the output from Wireshark it seams that the DNS of my dsl modem can not resolve the connection to sipgate, but that happends also before, or? The PBX send REGISTER to sipgate, but sipgate do not answer with 401 that the PBX can send REGISTER with the authorisation (in my opinion from the analysis of the Wireshark output). If it will help I can upload the log file of Wireshark. I think it is a little bit strange. Cheers zazi

Hi guys, I found the solution. I forwarded the call to 0814@[iP of my SpeechServer];transport=tcp. This construct isn't good for PBXNSIP in that field (as you maybe can see the: after "redirect to" in the log above is nothing before the system begins to check agains the dial plan). So I changed it just to 0814 (the number of my Speech Server application, but it this it not really relevant, because you have to add this again in your dial plan). Now it like to validate against +49814 (it adds automatically the country prefix). So you have to add it again to the dial plan and for the replacement I took now 0814 and it connects to my Speech Server application. Unfortunatelly it do not reconizes my voice input - maybe my telephone is so bad or something else. Hope that will maybe help other ones. Cheers zazi

Hello everybody, I tried hard to get the following setup to work: I have Speech Server application, where I can successfully connect from softphone (X-Lite client), which is connected to PBXNSIP. Now I like to phone from a hardware telephone. So I sign an account by Sipgate and registered it as a "SIP registration" trunk on my PBXNSIP system. Furthermore, I created an extension with the SIP-ID of Sipgate, that the incomming call has an trunk as "start point". In the Dial plan setup I routed the SIP-ID to the number of my Speech Server application, but everything I got until now is the mailbox (which is now disabled) and that the service is temporarily not available. So I think it should be something with the routing. I thought the configuration should be quite similar to that one of my softphone. Here I have log snippets of both connections: 1. from the softphone: [7] 2008/11/23 21:30:05: SIP Rx udp:192.168.1.111:53766: REGISTER sip:192.168.1.111:7060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-0f078d3b3f35d84b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766;rinstance=3d8ae1d0407da718> To: "4321"<sip:4321@192.168.1.111:7060> From: "4321"<sip:4321@192.168.1.111:7060>;tag=3b379f04 Call-ID: OGEzN2JlY2RmNzQ5ZDMwNjhkN2MzYmU4M2FiNmEyMDE. CSeq: 7 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="4321",realm="192.168.1.111",nonce="7feed25b310955128f9aeb885c8998ea",uri="sip:192.168.1.111:7060",response="ad9353ae5eda61f1dc7de1763e90ff53",algorithm=MD5 Content-Length: 0 [9] 2008/11/23 21:30:05: Resolve 331: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:05: SIP Tx udp:192.168.1.111:53766: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-0f078d3b3f35d84b-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=3b379f04 To: "4321" <sip:4321@192.168.1.111:7060>;tag=75f488e4b9 Call-ID: OGEzN2JlY2RmNzQ5ZDMwNjhkN2MzYmU4M2FiNmEyMDE. CSeq: 7 REGISTER Contact: <sip:4321@192.168.1.111:53766;rinstance=3d8ae1d0407da718>;expires=28 Content-Length: 0 [7] 2008/11/23 21:30:08: SIP Rx udp:192.168.1.111:53766: PUBLISH sip:4321@192.168.1.111:7060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-c030d1640b747368-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766> To: "4321"<sip:4321@192.168.1.111:7060> From: "4321"<sip:4321@192.168.1.111:7060>;tag=0d184055 Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk. CSeq: 2 PUBLISH Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/pidf+xml SIP-If-Match: b9g4pa User-Agent: X-Lite release 1100l stamp 47546 Event: presence Content-Length: 450 <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:4321@192.16'><tuple id='t6b2be110'><status><basic>open</basic></status></tuple><dm:person id='t6b2be110'><rpid:activities><rpid:on-the-phone/></rpid:activities><dm:note>On the Phone</dm:note></dm:person></presence> [9] 2008/11/23 21:30:08: Resolve 332: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:08: SIP Tx udp:192.168.1.111:53766: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-c030d1640b747368-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=0d184055 To: "4321" <sip:4321@192.168.1.111:7060>;tag=55447f1982 Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk. CSeq: 2 PUBLISH SIP-ETag: b9g4pa Expires: 3600 Content-Length: 0 [5] 2008/11/23 21:30:08: SIP port accept from 192.168.1.111:1421 [7] 2008/11/23 21:30:10: SIP Rx udp:192.168.1.111:53766: INVITE sip:0814@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766> To: "PizzaOrder"<sip:0814@192.168.1.111> From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 484 v=0 o=- 4 2 IN IP4 192.168.1.111 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.111 t=0 0 m=audio 53768 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101 a=alt:1 1 : IF9wAIIz Urysamk7 192.168.1.111 53768 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:102 L16/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [9] 2008/11/23 21:30:10: UDP: Opening socket on port 51420 [9] 2008/11/23 21:30:10: UDP: Opening socket on port 51421 [8] 2008/11/23 21:30:10: Could not find a trunk (3 trunks) [8] 2008/11/23 21:30:10: Using outbound proxy sip:192.168.1.111:53766;transport=udp because UDP packet source did not match the via header [9] 2008/11/23 21:30:10: Resolve 333: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09 To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 1 INVITE Content-Length: 0 [9] 2008/11/23 21:30:10: Resolve 334: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09 To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 1 INVITE User-Agent: pbxnsip-PBX/3.0.1.3023 WWW-Authenticate: Digest realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",domain="sip:0814@192.168.1.111",algorithm=MD5 Content-Length: 0 [7] 2008/11/23 21:30:10: SIP Rx udp:192.168.1.111:53766: ACK sip:0814@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d254ed1b001ce96e-1---d8754z-;rport To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 1 ACK Content-Length: 0 [7] 2008/11/23 21:30:10: SIP Rx udp:192.168.1.111:53766: INVITE sip:0814@192.168.1.111 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766> To: "PizzaOrder"<sip:0814@192.168.1.111> From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="4321",realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",uri="sip:0814@192.168.1.111",response="c02ce56f89a56e0498b5d0832739958a",algorithm=MD5 Content-Length: 484 v=0 o=- 4 2 IN IP4 192.168.1.111 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.111 t=0 0 m=audio 53768 RTP/AVP 107 119 100 106 6 0 97 105 98 8 102 3 5 101 a=alt:1 1 : IF9wAIIz Urysamk7 192.168.1.111 53768 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:102 L16/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv [8] 2008/11/23 21:30:10: Tagging request with existing tag [6] 2008/11/23 21:30:10: Sending RTP for Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.#ffc0e8c2c1 to 192.168.1.111:53768 [9] 2008/11/23 21:30:10: Resolve 335: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766: SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09 To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 2 INVITE Content-Length: 0 [9] 2008/11/23 21:30:10: Dialplan: Evaluating !^(4321)@.*!sip:0814@\r;user=phone!i against 0814@192.168.1.111 [9] 2008/11/23 21:30:10: Dialplan: Evaluating !^(0814)@.*!sip:\1@\r;user=phone!i against 0814@192.168.1.111 [5] 2008/11/23 21:30:10: Dialplan PizzaOrder: Match 0814@192.168.1.111 to <sip:0814@192.168.1.111;user=phone> on trunk MSSpeechServer [5] 2008/11/23 21:30:10: Charge user 4321 for redirecting calls [8] 2008/11/23 21:30:10: Play audio_moh/noise.wav [9] 2008/11/23 21:30:10: UDP: Opening socket on port 59724 [9] 2008/11/23 21:30:10: UDP: Opening socket on port 59725 [9] 2008/11/23 21:30:10: Resolve 336: url sip:192.168.1.111:15060;transport=tcp [9] 2008/11/23 21:30:10: Resolve 336: a tcp 192.168.1.111 15060 [9] 2008/11/23 21:30:10: Resolve 336: tcp 192.168.1.111 15060 [7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:15060: INVITE sip:0814@192.168.1.111;user=phone SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941 To: <sip:0814@192.168.1.111;user=phone> Call-ID: 8493d05b@pbx CSeq: 5041 INVITE Max-Forwards: 70 Contact: <sip:100@127.0.0.1:1423;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Type: application/sdp Content-Length: 284 v=0 o=- 63976 63976 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 59724 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:15060: SIP/2.0 100 Trying FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941 TO: <sip:0814@192.168.1.111;user=phone> CSEQ: 5041 INVITE CALL-ID: 8493d05b@pbx VIA: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport CONTENT-LENGTH: 0 [9] 2008/11/23 21:30:10: Resolve 337: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766: SIP/2.0 183 Ringing Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09 To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 2 INVITE Contact: <sip:4321@127.0.0.1:7060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Type: application/sdp Content-Length: 233 v=0 o=- 15820 15820 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 51420 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:15060: SIP/2.0 302 Moved Temporarily FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941 TO: <sip:0814@192.168.1.111;user=phone>;tag=3899c73dd5 CSEQ: 5041 INVITE CALL-ID: 8493d05b@pbx VIA: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport CONTACT: <sip:0814@192.168.1.111:6060;user=phone;transport=Tcp;maddr=192.168.1.111;x-mss-call-id=8493d05b%40pbx> CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/11/23 21:30:10: Call 8493d05b@pbx#10941: Clear last INVITE [9] 2008/11/23 21:30:10: Resolve 338: url sip:192.168.1.111:15060;transport=tcp [9] 2008/11/23 21:30:10: Resolve 338: a tcp 192.168.1.111 15060 [9] 2008/11/23 21:30:10: Resolve 338: tcp 192.168.1.111 15060 [7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:15060: ACK sip:0814@192.168.1.111;user=phone SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:1423;branch=z9hG4bK-a79b4caf22387f8f9ab1bc30cb7925c7;rport From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941 To: <sip:0814@192.168.1.111;user=phone>;tag=3899c73dd5 Call-ID: 8493d05b@pbx CSeq: 5041 ACK Max-Forwards: 70 Contact: <sip:100@127.0.0.1:1423;transport=tcp> Content-Length: 0 [5] 2008/11/23 21:30:10: Redirecting call [9] 2008/11/23 21:30:10: Resolve 339: aaaa tcp 192.168.1.111 6060 [9] 2008/11/23 21:30:10: Resolve 339: a tcp 192.168.1.111 6060 [9] 2008/11/23 21:30:10: Resolve 339: tcp 192.168.1.111 6060 [7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:6060: INVITE sip:0814@192.168.1.111:6060;user=phone;transport=Tcp;maddr=192.168.1.111;x-mss-call-id=8493d05b%40pbx SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941 To: <sip:0814@192.168.1.111;user=phone> Call-ID: 8493d05b@pbx CSeq: 5042 INVITE Max-Forwards: 70 Contact: <sip:100@127.0.0.1:1425;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Type: application/sdp Content-Length: 284 v=0 o=- 63976 63976 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 59724 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [6] 2008/11/23 21:30:10: Sending RTP for Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.#ffc0e8c2c1 to 127.0.0.1:53768 [7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:6060: SIP/2.0 100 Trying FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941 TO: <sip:0814@192.168.1.111;user=phone> CSEQ: 5042 INVITE CALL-ID: 8493d05b@pbx VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport CONTENT-LENGTH: 0 [7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:6060: SIP/2.0 180 Ringing FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941 TO: <sip:0814@192.168.1.111;user=phone>;epid=3E41C36034;tag=392e214f18 CSEQ: 5042 INVITE CALL-ID: 8493d05b@pbx VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [8] 2008/11/23 21:30:10: Play audio_en/ringback.wav [6] 2008/11/23 21:30:10: Sending RTP for 8493d05b@pbx#10941 to 192.168.1.111:13440 [7] 2008/11/23 21:30:10: SIP Rx tcp:192.168.1.111:6060: SIP/2.0 200 OK FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941 TO: <sip:0814@192.168.1.111;user=phone>;epid=3E41C36034;tag=392e214f18 CSEQ: 5042 INVITE CALL-ID: 8493d05b@pbx VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-0f4d1da0b6d4b796600a6621b56f3977;rport CONTACT: <sip:LEPPI01:6060;transport=Tcp;maddr=192.168.1.111>;automata CONTENT-LENGTH: 196 CONTENT-TYPE: application/sdp ALLOW: UPDATE SERVER: RTCC/3.0.0.0 ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 192.168.1.111 s=Microsoft Speech Server session c=IN IP4 192.168.1.111 t=0 0 m=audio 13440 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2008/11/23 21:30:10: Call 8493d05b@pbx#10941: Clear last INVITE [7] 2008/11/23 21:30:10: Set packet length to 20 [9] 2008/11/23 21:30:10: Resolve 340: aaaa tcp 192.168.1.111 6060 [9] 2008/11/23 21:30:10: Resolve 340: a tcp 192.168.1.111 6060 [9] 2008/11/23 21:30:10: Resolve 340: tcp 192.168.1.111 6060 [7] 2008/11/23 21:30:10: SIP Tx tcp:192.168.1.111:6060: ACK sip:LEPPI01:6060;transport=Tcp;maddr=192.168.1.111 SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-fee5e5c4eb71384086a295ec5034ac76;rport From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941 To: <sip:0814@192.168.1.111;user=phone>;tag=392e214f18 Call-ID: 8493d05b@pbx CSeq: 5042 ACK Max-Forwards: 70 Contact: <sip:100@127.0.0.1:1425;transport=tcp> Content-Length: 0 [7] 2008/11/23 21:30:10: Determine pass-through mode after receiving response [9] 2008/11/23 21:30:10: Resolve 341: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:10: SIP Tx udp:192.168.1.111:53766: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-b2608d762672484e-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09 To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 2 INVITE Contact: <sip:4321@127.0.0.1:7060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Content-Type: application/sdp Content-Length: 233 v=0 o=- 15820 15820 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 51420 RTP/AVP 0 8 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/11/23 21:30:10: SIP Rx udp:127.0.0.1:53766: ACK sip:4321@127.0.0.1:7060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-f90fa7068f2f5109-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766> To: "PizzaOrder"<sip:0814@192.168.1.111>;tag=ffc0e8c2c1 From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 2 ACK User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="4321",realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",uri="sip:0814@192.168.1.111",response="c02ce56f89a56e0498b5d0832739958a",algorithm=MD5 Content-Length: 0 [7] 2008/11/23 21:30:10: 8493d05b@pbx#10941: RTP pass-through mode [7] 2008/11/23 21:30:10: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA.#ffc0e8c2c1: RTP pass-through mode [7] 2008/11/23 21:30:13: SIP Rx udp:127.0.0.1:53766: BYE sip:4321@127.0.0.1:7060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d6017e10377e7a0d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766> To: "PizzaOrder"<sip:0814@192.168.1.111>;tag=ffc0e8c2c1 From: "4321"<sip:4321@192.168.1.111:7060>;tag=18323e09 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 3 BYE User-Agent: X-Lite release 1100l stamp 47546 Authorization: Digest username="4321",realm="192.168.1.111",nonce="34e0ecd53dc31bfd7464f7b7a7942fd2",uri="sip:4321@127.0.0.1:7060",response="044d4a36644d99fadc4e61330d41253f",algorithm=MD5 Reason: SIP;description="User Hung Up" Content-Length: 0 [9] 2008/11/23 21:30:13: Resolve 342: aaaa udp 127.0.0.1 53766 [9] 2008/11/23 21:30:13: Resolve 342: a udp 127.0.0.1 53766 [9] 2008/11/23 21:30:13: Resolve 342: udp 127.0.0.1 53766 [7] 2008/11/23 21:30:13: SIP Tx udp:127.0.0.1:53766: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-d6017e10377e7a0d-1---d8754z-;rport=53766 From: "4321" <sip:4321@192.168.1.111:7060>;tag=18323e09 To: "PizzaOrder" <sip:0814@192.168.1.111>;tag=ffc0e8c2c1 Call-ID: Y2RiZmM1OWU0ZmNlYWFhZmExMjA3ZDdhYWI2OGQ4MTA. CSeq: 3 BYE Contact: <sip:4321@127.0.0.1:7060> User-Agent: pbxnsip-PBX/3.0.1.3023 RTP-RxStat: Dur=3,Pkt=146,Oct=25112,Underun=0 RTP-TxStat: Dur=3,Pkt=147,Oct=25284 Content-Length: 0 [7] 2008/11/23 21:30:13: 8493d05b@pbx#10941: Media-aware pass-through mode [7] 2008/11/23 21:30:13: Other Ports: 1 [7] 2008/11/23 21:30:13: Call Port: 8493d05b@pbx#10941 [7] 2008/11/23 21:30:13: SIP Rx udp:192.168.1.111:53766: PUBLISH sip:4321@192.168.1.111:7060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-114af4181f09301a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4321@192.168.1.111:53766> To: "4321"<sip:4321@192.168.1.111:7060> From: "4321"<sip:4321@192.168.1.111:7060>;tag=0d184055 Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk. CSeq: 3 PUBLISH Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/pidf+xml SIP-If-Match: b9g4pa User-Agent: X-Lite release 1100l stamp 47546 Event: presence Content-Length: 414 <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:4321@192.16'><tuple id='t6b2be110'><status><basic>open</basic></status></tuple><dm:person id='t6b2be110'><rpid:activities><rpid:unknown/></rpid:activities></dm:person></presence> [9] 2008/11/23 21:30:13: Resolve 343: udp 192.168.1.111 53766 [7] 2008/11/23 21:30:13: SIP Tx udp:192.168.1.111:53766: SIP/2.0 200 Ok Via: SIP/2.0/UDP 127.0.0.1:53766;branch=z9hG4bK-d8754z-114af4181f09301a-1---d8754z-;rport=53766;received=192.168.1.111 From: "4321" <sip:4321@192.168.1.111:7060>;tag=0d184055 To: "4321" <sip:4321@192.168.1.111:7060>;tag=55447f1982 Call-ID: NmRmZGE5MmUxOWUxMmY5ZWE2Mzg2MWJmNjNiYzIxODk. CSeq: 3 PUBLISH SIP-ETag: b9g4pa Expires: 3600 Content-Length: 0 [9] 2008/11/23 21:30:13: Resolve 344: aaaa tcp 192.168.1.111 6060 [9] 2008/11/23 21:30:13: Resolve 344: a tcp 192.168.1.111 6060 [9] 2008/11/23 21:30:13: Resolve 344: tcp 192.168.1.111 6060 [7] 2008/11/23 21:30:13: SIP Tx tcp:192.168.1.111:6060: BYE sip:LEPPI01:6060;transport=Tcp;maddr=192.168.1.111 SIP/2.0 Via: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-d0cd32ed84b1bf136da63e679df6f801;rport From: "zazi" <sip:100@192.168.1.111;user=phone>;tag=10941 To: <sip:0814@192.168.1.111;user=phone>;tag=392e214f18 Call-ID: 8493d05b@pbx CSeq: 5043 BYE Max-Forwards: 70 Contact: <sip:100@127.0.0.1:1425;transport=tcp> RTP-RxStat: Dur=3,Pkt=133,Oct=22876,Underun=0 RTP-TxStat: Dur=3,Pkt=129,Oct=22188 Content-Length: 0 [7] 2008/11/23 21:30:13: SIP Rx tcp:192.168.1.111:6060: SIP/2.0 200 OK FROM: "zazi"<sip:100@192.168.1.111;user=phone>;tag=10941 TO: <sip:0814@192.168.1.111;user=phone>;tag=392e214f18;epid=3E41C36034 CSEQ: 5043 BYE CALL-ID: 8493d05b@pbx VIA: SIP/2.0/TCP 127.0.0.1:1425;branch=z9hG4bK-d0cd32ed84b1bf136da63e679df6f801;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.0.0.0 [7] 2008/11/23 21:30:13: Call 8493d05b@pbx#10941: Clear last request [5] 2008/11/23 21:30:13: BYE Response: Terminate 8493d05b@pbx ================================================================== 53766 is the port where the softphone client is listening 7060 is the tcp port of my PBXNSIP system 6060 is the port of my Speech Server application 100 is the Trunk ANI of the Speech Server SIP gateway 4321 is the number/name of my softphone client, which is an extension account at my PBXNSIP system PizzaOrder is the name of my dial plan (because it should connect to the PizzaOrder tutorial application from MS Speech Server) ================================================================== 2. from my hardphone: [7] 2008/11/23 20:41:00: SIP Rx udp:217.10.79.9:5060: INVITE sip:[my Sipgate SIP-ID]@192.168.2.100:7060;transport=udp;line=a87ff679 SIP/2.0 Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> Record-Route: <sip:172.20.40.2;lr=on> Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060 From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d To: <sip:[my Sipgate phone number with country prefix]@sipgate.de> Contact: <sip:[my real hardphone number]@217.10.67.5> Call-ID: 7992bc44273786571088032e273db69c@sipgate.de CSeq: 102 INVITE Max-Forwards: 67 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 408 v=0 o=root 24764 24764 IN IP4 217.10.67.5 s=session c=IN IP4 217.10.67.5 t=0 0 m=audio 11354 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [9] 2008/11/23 20:41:00: UDP: Opening socket on port 54212 [9] 2008/11/23 20:41:00: UDP: Opening socket on port 54213 [5] 2008/11/23 20:41:00: Identify trunk (line match) 4 [9] 2008/11/23 20:41:00: Resolve 221: aaaa udp 217.10.79.9 5060 [9] 2008/11/23 20:41:00: Resolve 221: a udp 217.10.79.9 5060 [9] 2008/11/23 20:41:00: Resolve 221: udp 217.10.79.9 5060 [7] 2008/11/23 20:41:00: SIP Tx udp:217.10.79.9:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060 Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> Record-Route: <sip:172.20.40.2;lr=on> Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1 Call-ID: 7992bc44273786571088032e273db69c@sipgate.de CSeq: 102 INVITE Content-Length: 0 [7] 2008/11/23 20:41:00: Set packet length to 20 [6] 2008/11/23 20:41:00: Sending RTP for 7992bc44273786571088032e273db69c@sipgate.de#0d0d85e1a1 to 217.10.67.5:11354 [5] 2008/11/23 20:41:00: Trunk Sipgate sends call to [my Sipgate SIP-ID] in domain pbx.company.com [7] 2008/11/23 20:41:00: Attendant: Calling extension [my Sipgate SIP-ID] [5] 2008/11/23 20:41:00: Attendant: Redirect to [9] 2008/11/23 20:41:00: Dialplan: Evaluating !^(4321)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com [9] 2008/11/23 20:41:00: Dialplan: Evaluating !^(0814)@.*!sip:\1@\r;user=phone!i against sip:pbx.company.com@pbx.company.com [9] 2008/11/23 20:41:00: Dialplan: Evaluating !^9181([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com [9] 2008/11/23 20:41:00: Dialplan: Evaluating !^0351([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com [9] 2008/11/23 20:41:00: Dialplan: Evaluating !^9181([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com [9] 2008/11/23 20:41:00: Dialplan: Evaluating !^0049351([0-9]*)@.*!sip:0814@\r;user=phone!i against sip:pbx.company.com@pbx.company.com [7] 2008/11/23 20:41:00: Set packet length to 20 [9] 2008/11/23 20:41:00: Resolve 222: aaaa udp 217.10.79.9 5060 [9] 2008/11/23 20:41:00: Resolve 222: a udp 217.10.79.9 5060 [9] 2008/11/23 20:41:00: Resolve 222: udp 217.10.79.9 5060 [7] 2008/11/23 20:41:00: SIP Tx udp:217.10.79.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060 Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> Record-Route: <sip:172.20.40.2;lr=on> Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1 Call-ID: 7992bc44273786571088032e273db69c@sipgate.de CSeq: 102 INVITE Contact: <sip:[my Sipgate SIP-ID]@192.168.1.111:7060;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/3.0.1.3023 Content-Length: 0 [9] 2008/11/23 20:41:00: Resolve 223: aaaa udp 217.10.79.9 5060 [9] 2008/11/23 20:41:00: Resolve 223: a udp 217.10.79.9 5060 [9] 2008/11/23 20:41:00: Resolve 223: udp 217.10.79.9 5060 [7] 2008/11/23 20:41:00: SIP Tx udp:217.10.79.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060 Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> Record-Route: <sip:172.20.40.2;lr=on> Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1 Call-ID: 7992bc44273786571088032e273db69c@sipgate.de CSeq: 102 INVITE Contact: <sip:[my Sipgate SIP-ID]@192.168.1.111:7060;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/3.0.1.3023 Content-Length: 0 [7] 2008/11/23 20:41:00: SIP Tr udp:217.10.79.9:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bKf4c8.91cbc5e.0 Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK54fb519d Via: SIP/2.0/UDP 217.10.67.5:5060;branch=z9hG4bK54fb519d;rport=5060 Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> Record-Route: <sip:172.20.40.2;lr=on> Record-Route: <sip:217.10.79.9;lr=on;ftag=as1a76684d> From: "[my real hardphone number]" <sip:[my real hardphone number]@sipgate.de>;tag=as1a76684d To: <sip:[my Sipgate phone number with country prefix]@sipgate.de>;tag=0d0d85e1a1 Call-ID: 7992bc44273786571088032e273db69c@sipgate.de CSeq: 102 INVITE Contact: <sip:[my Sipgate SIP-ID]@192.168.1.111:7060;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/3.0.1.3023 Content-Length: 0 ============================================================ At last, I thought it has to something with the country code, because the Sipgate phone number comes along with the full country prefix, but I also add that to my dial plan. As you can see in the last log above I tried different routing, without any positive result. A bad side effect is after one call via the Sipgate SIP gateway the registration get and I have to reboot my system. Thanks a lot for any help. Cheers zazi

Okay, I figured out the task, which is responsible for kicking my registration. Therefore, I have to explain the situation of my development/ test state. My test goal is, to phone a IVR hosted on a MS Speech Server which is connect with the PBXNSIP. The first test: calling the application from a softphone, which is connected to PBXNSIP succeed. Now it is time for establishing the routes from the sipgate phone number to the speech application (but I do not really know how to that ). So I tried different configuration and test them with a phone call. Always after that phone calls, the sipgate registration get lost. Before I initiate a call, everything is fine. I checked that out with Wireshark (after 30 seconds the registration refreshes successfully). So it depend more or less indirectly on the sipgate registration. cheers zazi

Hm, I tried out the 30 seconds variant, but after the first refresh the registration get lost. I disabled my firewall. It could just something to do with my router. I can check the system tomorrow somewhere else. However, I can only mentioned that a re-registration in 3CX works. Nevertheless, thanks a lot for help. cheers zazi

Yeah, I checked out again my other test PBX (3CX Phone System). There the registration will be refreshed after my configurated period of time successfully. So maybe it has something to with my configuration at PBXNSIP, or I don't know. 3CX make use of the STUN server of Sipgate. Unfortunatelly, this isn't any more available at PBXNSIP. The big disadvantage of 3CX is that it does not work together with MS Speech Server 2007. PBXNSIP does it, fortunatelly. So here is the configuration of my SIP registration in PBXNSIP: Name: Type: SIP Registration Direction Inbound and outbound -------------------------------------------------------------------------------- Display Name: [my Sipgate SIP-ID] Account: [my Sipgate SIP-ID] Domain: sipgate.de Username: [my Sipgate SIP-ID] Password: [my Sipgate SIP-password] Password (repeat): [my Sipgate SIP-password] Outbound Proxy: sipgate.de CO Lines: - Dialog Permissions: - Codec Preference: - Proposed Duration (s): 3600 Keepalive Time: 600 Send email on status change: yes Strict RTP Routing: no Avoid RFC4122 (UUID): no Accept Redirect: yes Interpret SIP URI always as telephone number: yes -------------------------------------------------------------------------------- Prefix: - Global: no Trunk ANI: [my external phone number of my Sipgate account] Remote Party/Privacy Indication: Remote-Party-ID Failover Behavior: No failover Is Secure: no ICID (RFC 3455): - -------------------------------------------------------------------------------- Send call to extension: - Assume that call comes from user: - Ringback: Media checked Hope that will help to analyse the problem. cheers zazi

As I mentioned, here is the excerpt of the log, when it connects successful. [9] 2008/11/23 16:11:31: Resolve 3: url sip:sipgate.de [9] 2008/11/23 16:11:31: Resolve 3: naptr sipgate.de [8] 2008/11/23 16:11:32: DNS: Add dns_naptr sipgate.de (ttl=60) [9] 2008/11/23 16:11:32: Resolve 3: naptr sipgate.de [9] 2008/11/23 16:11:32: Resolve 3: srv tls _sips._tcp.sipgate.de [8] 2008/11/23 16:11:34: DNS: Add dns_srv _sips._tcp.sipgate.de (ttl=60) [9] 2008/11/23 16:11:34: Resolve 3: srv tls _sips._tcp.sipgate.de [9] 2008/11/23 16:11:34: Resolve 3: srv tcp _sip._tcp.sipgate.de [8] 2008/11/23 16:11:35: DNS: Add dns_srv _sip._tcp.sipgate.de (ttl=60) [9] 2008/11/23 16:11:35: Resolve 3: srv tcp _sip._tcp.sipgate.de [9] 2008/11/23 16:11:35: Resolve 3: srv udp _sip._udp.sipgate.de [8] 2008/11/23 16:11:35: DNS: Add dns_srv _sip._udp.sipgate.de 0 0 sipgate.de 5060 (ttl=19219) [9] 2008/11/23 16:11:35: Resolve 3: srv udp _sip._udp.sipgate.de [9] 2008/11/23 16:11:35: Resolve 3: a udp sipgate.de 5060 [8] 2008/11/23 16:11:35: DNS: Add dns_a sipgate.de 217.10.79.9 (ttl=7862) [9] 2008/11/23 16:11:35: Resolve 3: a udp sipgate.de 5060 [9] 2008/11/23 16:11:35: Resolve 3: udp 217.10.79.9 5060 [9] 2008/11/23 16:11:35: Resolve 3: a udp sipgate.de 5060 [9] 2008/11/23 16:11:35: Resolve 3: udp 217.10.79.9 5060 [8] 2008/11/23 16:11:35: Trunk 4 (Sipgate) has outbound proxy udp:217.10.79.9:5060 [9] 2008/11/23 16:11:35: Resolve 4: url sip:sipgate.de [9] 2008/11/23 16:11:35: Resolve 4: naptr sipgate.de [9] 2008/11/23 16:11:35: Resolve 4: srv tls _sips._tcp.sipgate.de [9] 2008/11/23 16:11:35: Resolve 4: srv tcp _sip._tcp.sipgate.de [9] 2008/11/23 16:11:35: Resolve 4: srv udp _sip._udp.sipgate.de [9] 2008/11/23 16:11:35: Resolve 4: a udp sipgate.de 5060 [9] 2008/11/23 16:11:35: Resolve 4: udp 217.10.79.9 5060 [7] 2008/11/23 16:11:35: SIP Tx udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-9cf51e6e48f77b785723e86e1dec4449;rport From: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de>;tag=46466 To: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de> Call-ID: ln1f246s@pbx CSeq: 26009 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP_ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Expires: 3600 Content-Length: 0 [7] 2008/11/23 16:11:35: SIP Rx udp:217.10.79.9:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.100:7060;branch=z9hG4bK-9cf51e6e48f77b785723e86e1dec4449;rport=7060 From: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de>;tag=46466 To: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de>;tag=8367f0f887e3954243ec30fa0f5db288.6e3c Call-ID: ln1f246s@pbx CSeq: 26009 REGISTER WWW-Authenticate: Digest realm="sipgate.de", nonce="492973d31fc3982ac267611b4db6ff3593c9b7d1" Content-Length: 0 [8] 2008/11/23 16:11:35: Answer challenge with username [my Sipgate SIP_ID] [9] 2008/11/23 16:11:35: Resolve 5: udp 217.10.79.9 5060 udp:1 [7] 2008/11/23 16:11:35: SIP Tx udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-200fb471786ba657704995c99944ff82;rport From: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de>;tag=46466 To: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de> Call-ID: ln1f246s@pbx CSeq: 26010 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP_ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Authorization: Digest realm="sipgate.de",nonce="492973d31fc3982ac267611b4db6ff3593c9b7d1",response="5cb893f95205f2e7402de3ce0e0133e7",username="[my Sipgate SIP_ID]",uri="sip:sipgate.de",algorithm=MD5 Expires: 3600 Content-Length: 0 [9] 2008/11/23 16:11:35: Message repetition, packet dropped [7] 2008/11/23 16:11:35: SIP Rx udp:217.10.79.9:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-200fb471786ba657704995c99944ff82;rport=7060 From: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de>;tag=46466 To: "[my Sipgate SIP_ID]" <sip:[my Sipgate SIP_ID]@sipgate.de>;tag=8367f0f887e3954243ec30fa0f5db288.4e5b Call-ID: ln1f246s@pbx CSeq: 26010 REGISTER Contact: <sip:[my Sipgate SIP_ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;expires=600 Content-Length: 0 [2] 2008/11/23 16:11:35: Trunk status Sipgate (4) changed to "200 OK" (Refresh interval 600 seconds) [8] 2008/11/23 16:11:36: No from address for sending text email [8] 2008/11/23 16:12:32: DNS: dns_naptr sipgate.de expired [8] 2008/11/23 16:12:34: DNS: dns_srv _sips._tcp.sipgate.de expired [8] 2008/11/23 16:12:35: DNS: dns_srv _sip._tcp.sipgate.de expired cheers zazi PS: I will try out the link you've posted.

Hello, I like to use my Sipgate account to register my telephone number for calls at PBXNSIP. Everthing works fine at the first time when I start my system and every service loads. After a while (maybe 3-5 minutes) the connection to the sipgate-server gets lost and when PBXNSIP tries to re-register it, it will fail after a while with the code 408 (timeout). So here is an excerpt from the log, when the systems tries to connect to sipgate: [9] 2008/11/23 15:47:52: Resolve 108: url sip:sipgate.de [9] 2008/11/23 15:47:52: Resolve 108: naptr sipgate.de [8] 2008/11/23 15:47:54: DNS: Add dns_naptr sipgate.de (ttl=60) [9] 2008/11/23 15:47:54: Resolve 108: naptr sipgate.de [9] 2008/11/23 15:47:54: Resolve 108: srv tls _sips._tcp.sipgate.de [8] 2008/11/23 15:47:55: DNS: Add dns_srv _sips._tcp.sipgate.de (ttl=60) [9] 2008/11/23 15:47:55: Resolve 108: srv tls _sips._tcp.sipgate.de [9] 2008/11/23 15:47:55: Resolve 108: srv tcp _sip._tcp.sipgate.de [8] 2008/11/23 15:47:57: DNS: Add dns_srv _sip._tcp.sipgate.de (ttl=60) [9] 2008/11/23 15:47:57: Resolve 108: srv tcp _sip._tcp.sipgate.de [9] 2008/11/23 15:47:57: Resolve 108: srv udp _sip._udp.sipgate.de [9] 2008/11/23 15:47:57: Resolve 108: a udp sipgate.de 5060 [9] 2008/11/23 15:47:57: Resolve 108: udp 217.10.79.9 5060 [9] 2008/11/23 15:47:57: Resolve 108: a udp sipgate.de 5060 [9] 2008/11/23 15:47:57: Resolve 108: udp 217.10.79.9 5060 [8] 2008/11/23 15:47:57: Trunk 4 (Sipgate) has outbound proxy udp:217.10.79.9:5060 [9] 2008/11/23 15:47:57: Resolve 109: url sip:sipgate.de [9] 2008/11/23 15:47:57: Resolve 109: naptr sipgate.de [9] 2008/11/23 15:47:57: Resolve 109: srv tls _sips._tcp.sipgate.de [9] 2008/11/23 15:47:57: Resolve 109: srv tcp _sip._tcp.sipgate.de [9] 2008/11/23 15:47:57: Resolve 109: srv udp _sip._udp.sipgate.de [9] 2008/11/23 15:47:57: Resolve 109: a udp sipgate.de 5060 [9] 2008/11/23 15:47:57: Resolve 109: udp 217.10.79.9 5060 [7] 2008/11/23 15:47:57: SIP Tx udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-13feb7b5bd0c5e02fe358ddd30aa471f;rport From: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de>;tag=26837 To: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de> Call-ID: 9n20y1a1@pbx CSeq: 13324 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP-ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Expires: 3600 Content-Length: 0 [7] 2008/11/23 15:47:57: SIP Tr udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-13feb7b5bd0c5e02fe358ddd30aa471f;rport From: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de>;tag=26837 To: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de> Call-ID: 9n20y1a1@pbx CSeq: 13324 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP-ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Expires: 3600 Content-Length: 0 [8] 2008/11/23 15:48:54: DNS: dns_naptr sipgate.de expired [8] 2008/11/23 15:48:55: DNS: dns_srv _sips._tcp.sipgate.de expired [8] 2008/11/23 15:48:57: DNS: dns_srv _sip._tcp.sipgate.de expired 9] 2008/11/23 15:49:27: Resolve 114: url sip:sipgate.de [9] 2008/11/23 15:49:27: Resolve 114: naptr sipgate.de [8] 2008/11/23 15:49:28: DNS: Add dns_naptr sipgate.de (ttl=60) [9] 2008/11/23 15:49:28: Resolve 114: naptr sipgate.de [9] 2008/11/23 15:49:28: Resolve 114: srv tls _sips._tcp.sipgate.de [8] 2008/11/23 15:49:30: DNS: Add dns_srv _sips._tcp.sipgate.de (ttl=60) [9] 2008/11/23 15:49:30: Resolve 114: srv tls _sips._tcp.sipgate.de [9] 2008/11/23 15:49:30: Resolve 114: srv tcp _sip._tcp.sipgate.de [8] 2008/11/23 15:49:31: DNS: Add dns_srv _sip._tcp.sipgate.de (ttl=60) [9] 2008/11/23 15:49:31: Resolve 114: srv tcp _sip._tcp.sipgate.de [9] 2008/11/23 15:49:31: Resolve 114: srv udp _sip._udp.sipgate.de [9] 2008/11/23 15:49:31: Resolve 114: a udp sipgate.de 5060 [9] 2008/11/23 15:49:31: Resolve 114: udp 217.10.79.9 5060 [9] 2008/11/23 15:49:31: Resolve 114: a udp sipgate.de 5060 [9] 2008/11/23 15:49:31: Resolve 114: udp 217.10.79.9 5060 [8] 2008/11/23 15:49:31: Trunk 4 (Sipgate) has outbound proxy udp:217.10.79.9:5060 [9] 2008/11/23 15:49:31: Resolve 115: url sip:sipgate.de [9] 2008/11/23 15:49:31: Resolve 115: naptr sipgate.de [9] 2008/11/23 15:49:31: Resolve 115: srv tls _sips._tcp.sipgate.de [9] 2008/11/23 15:49:31: Resolve 115: srv tcp _sip._tcp.sipgate.de [9] 2008/11/23 15:49:31: Resolve 115: srv udp _sip._udp.sipgate.de [9] 2008/11/23 15:49:31: Resolve 115: a udp sipgate.de 5060 [9] 2008/11/23 15:49:31: Resolve 115: udp 217.10.79.9 5060 [7] 2008/11/23 15:49:31: SIP Tx udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-fe1f9eda3c0b1f20fca77eecf3a00894;rport From: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de>;tag=26837 To: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de> Call-ID: 9n20y1a1@pbx CSeq: 13326 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP-ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Expires: 3600 Content-Length: 0 [7] 2008/11/23 15:49:32: SIP Tr udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-fe1f9eda3c0b1f20fca77eecf3a00894;rport From: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de>;tag=26837 To: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de> Call-ID: 9n20y1a1@pbx CSeq: 13326 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP-ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Expires: 3600 Content-Length: 0 [7] 2008/11/23 15:49:47: SIP Tr udp:217.10.79.9:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:7060;branch=z9hG4bK-fe1f9eda3c0b1f20fca77eecf3a00894;rport From: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de>;tag=26837 To: "[my Sipgate SIP-ID]" <sip:[my Sipgate SIP-ID]@sipgate.de> Call-ID: 9n20y1a1@pbx CSeq: 13326 REGISTER Max-Forwards: 70 Contact: <sip:[my Sipgate SIP-ID]@192.168.1.106:7060;transport=udp;line=a87ff679>;+sip.instance="<urn:uuid:c26897eb-2bf3-4c04-9bd3-99854322a6b9>" User-Agent: pbxnsip-PBX/3.0.1.3023 Expires: 3600 Content-Length: 0 [5] 2008/11/23 15:50:01: Registration on trunk 4 (Sipgate) failed. Retry in 60 seconds I replaced my sipgate sip-id with [my sigate sip-id]. 192.168.1.106 is the IP of the PBXNSIP system and 7060 the port I used for it. I can also get the snippets of successful registration from PBXNSIP with sipgate and also the configuration of that trunk. cheers zazi