Daniel Floeckinger

We are using our Fax connected to a Audiocodes MP-112 and a TELES VoipBox as PSTN Gateway. All t.38. The TELES unit uses the Audiocodes Chipset, so they are 100% compatible to each other. Before we changed to the Audiocodes, we were using a Linksys 2102 with t.38. This setup worked 95%.

Hi! We want to receive the following: When a caller calls the pbx from a trunk, no matter which extension (we have DDI here) he should first be redirected to a voice announcement. At the same time the dialed extension shluld start ringing. The voice announcement must allways start from the beginning. Every caller must hear it until the call is picked up by person or voicemail etc. Is this possible?

Yes indeed. It is very hard to configure a SNOM phone. PNP does not work at all. Sometimes, the phone finds the pbx and sometimes it does not. Yes, it "should" work. But in reality it is someting like playing Roulette. Dan.

The problem is still unsolved. Could you please respond, where to send the trace to?

Maybe because you are Administrator. I dont have this section, i am afraid.

Does not work :-( Have made a trace... dont have a upload button, so, where shall i send it?

Here u go... Both negotiate T38. But after that i can only see "malformed" packets in ethereal. The fax is 192.168.16.124, the Gateway 192.168.16.12, the pbx 192.168.16.110 [7] 2007/06/01 18:20:16: SIP Rx udp:192.168.16.124:5060: INVITE sip:0664771607997@192.168.16.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK2113019710;rport Route: <sip:192.168.16.110:5060;lr> From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110> Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 60 INVITE Contact: <sip:15@192.168.16.124:5060> Max-Forwards: 70 Supported: replaces, path User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Content-Length: 363 v=0 o=15 8000 8000 IN IP4 192.168.16.124 s=SIP Call c=IN IP4 192.168.16.124 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 103 102 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:103 G726-40/8000 a=rtpmap:102 G729E/8000 [0] 2007/06/01 18:20:16: UDP: Illegal port number [7] 2007/06/01 18:20:16: UDP: Opening socket on port 49966 [7] 2007/06/01 18:20:16: UDP: Opening socket on port 49967 [5] 2007/06/01 18:20:16: Identify trunk 5 [8] 2007/06/01 18:20:16: Resolve destination 43538: a udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Resolve destination 43538: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Send Packet 100 [7] 2007/06/01 18:20:16: SIP Tx udp:192.168.16.124:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK2113019710;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 60 INVITE Content-Length: 0 [8] 2007/06/01 18:20:16: Resolve destination 43539: a udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Resolve destination 43539: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Send Packet 401 [7] 2007/06/01 18:20:16: SIP Tx udp:192.168.16.124:5060: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK2113019710;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 60 INVITE User-Agent: FL-Connect IP-PBX/2.0.3.1715 WWW-Authenticate: Digest realm="192.168.16.110",nonce="69d3ec2d6c963ae7c3c5b49b6007f033",domain="sip:0664771607997@192.168.16.110",algorithm=MD5 Content-Length: 0 [7] 2007/06/01 18:20:16: SIP Rx udp:192.168.16.124:5060: ACK sip:0664771607997@192.168.16.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK2113019710;rport Route: <sip:192.168.16.110:5060;lr> From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 60 ACK Content-Length: 0 [7] 2007/06/01 18:20:16: SIP Rx udp:192.168.16.124:5060: INVITE sip:0664771607997@192.168.16.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport Route: <sip:192.168.16.110:5060;lr> From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110> Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.124:5060> Authorization: Digest username="15", realm="192.168.16.110", nonce="69d3ec2d6c963ae7c3c5b49b6007f033", uri="sip:0664771607997@192.168.16.110", response="ab9e1b2b4254ece8e78e910ebde5fdcc", algorithm=MD5 Max-Forwards: 70 Supported: replaces, path User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Content-Length: 363 v=0 o=15 8000 8000 IN IP4 192.168.16.124 s=SIP Call c=IN IP4 192.168.16.124 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 103 102 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:103 G726-40/8000 a=rtpmap:102 G729E/8000 [8] 2007/06/01 18:20:16: Tagging request with existing tag [8] 2007/06/01 18:20:16: Resolve destination 43541: a udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Resolve destination 43541: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Send Packet 100 [7] 2007/06/01 18:20:16: SIP Tx udp:192.168.16.124:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Content-Length: 0 [7] 2007/06/01 18:20:16: UDP: Opening socket on port 51042 [7] 2007/06/01 18:20:16: UDP: Opening socket on port 51043 [8] 2007/06/01 18:20:16: Resolve destination 43544: url sip:192.168.16.12 [8] 2007/06/01 18:20:16: Resolve destination 43544: udp 192.168.16.12 5060 [8] 2007/06/01 18:20:16: Send Packet INVITE [7] 2007/06/01 18:20:16: SIP Tx udp:192.168.16.12:5060: INVITE sip:0664771607997@192.168.16.12;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-2b23244bd8acf79a6618c117976b228a;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone> Call-ID: 841706e2@pbx CSeq: 17112 INVITE Max-Forwards: 70 Contact: <sip:sippbx@192.168.16.110:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Remote-Party-ID: "sippbx" <sip:sippbx@192.168.16.12>;party=calling;screen=yes Content-Type: application/sdp Content-Length: 296 v=0 o=- 11568 11568 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 51042 RTP/AVP 0 8 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=sendrecv [8] 2007/06/01 18:20:16: Resolve destination 43545: a udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Resolve destination 43545: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:16: Send Packet 183 [7] 2007/06/01 18:20:16: SIP Tx udp:192.168.16.124:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 152 v=0 o=- 29725 29725 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 49966 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 a=sendrecv [7] 2007/06/01 18:20:16: SIP Rx udp:192.168.16.12:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-2b23244bd8acf79a6618c117976b228a;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone> Call-ID: 841706e2@pbx CSeq: 17112 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER User-Agent: TELES.VoIPBOX 13.0b 885 Content-Length: 0 7] 2007/06/01 18:20:17: SIP Tr udp:192.168.16.124:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 152 v=0 o=- 29725 29725 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 49966 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 a=sendrecv [8] 2007/06/01 18:20:18: Send Packet 183 [7] 2007/06/01 18:20:18: SIP Tr udp:192.168.16.124:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 152 v=0 o=- 29725 29725 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 49966 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 a=sendrecv [8] 2007/06/01 18:20:20: Send Packet 183 [7] 2007/06/01 18:20:20: SIP Tr udp:192.168.16.124:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 152 v=0 o=- 29725 29725 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 49966 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 a=sendrecv [7] 2007/06/01 18:20:22: SIP Rx udp:192.168.16.12:5060: SIP/2.0 183 Session progress Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-2b23244bd8acf79a6618c117976b228a;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 Call-ID: 841706e2@pbx CSeq: 17112 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: <sip:0664771607997@192.168.16.12> User-Agent: TELES.VoIPBOX 13.0b 885 Content-Type: application/sdp Content-Length: 188 v=0 o=- 277 1 IN IP4 192.168.16.12 s=- c=IN IP4 192.168.16.12 t=0 0 m=audio 29000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:40 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [8] 2007/06/01 18:20:22: Resolve destination 43547: url sip:35@192.168.16.123:2054;line=4ym9kfvn [8] 2007/06/01 18:20:22: Resolve destination 43547: udp 192.168.16.123 2054 [8] 2007/06/01 18:20:22: Send Packet NOTIFY [7] 2007/06/01 18:20:22: SIP Tx udp:192.168.16.123:2054: NOTIFY sip:35@192.168.16.123:2054;line=4ym9kfvn SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-c5aa0cd3873a1e0a271234f3bed12af3;rport From: <sip:15@localhost;user=phone>;tag=003530fce3 To: <sip:35@localhost>;tag=qdifvavd0l Call-ID: 3c26700e3b25-np70hd4mtm3d CSeq: 1372 NOTIFY Max-Forwards: 70 Contact: <sip:192.168.16.110:5060;transport=udp> Event: dialog Subscription-State: active;expires=66 Content-Type: application/dialog-info+xml Content-Length: 455 [8] 2007/06/01 18:20:24: Send Packet 183 [7] 2007/06/01 18:20:24: SIP Tr udp:192.168.16.124:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 152 v=0 o=- 29725 29725 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 49966 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 a=sendrecv [7] 2007/06/01 18:20:24: SIP Rx udp:192.168.16.12:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-2b23244bd8acf79a6618c117976b228a;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 Call-ID: 841706e2@pbx CSeq: 17112 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: <sip:0664771607997@192.168.16.12> User-Agent: TELES.VoIPBOX 13.0b 885 Content-Type: application/sdp Content-Length: 188 v=0 o=- 277 1 IN IP4 192.168.16.12 s=- c=IN IP4 192.168.16.12 t=0 0 m=audio 29000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:40 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [7] 2007/06/01 18:20:24: SIP Rx udp:192.168.16.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-2b23244bd8acf79a6618c117976b228a;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 Call-ID: 841706e2@pbx CSeq: 17112 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: <sip:0664771607997@192.168.16.12> User-Agent: TELES.VoIPBOX 13.0b 885 Content-Type: application/sdp Content-Length: 188 v=0 o=- 277 1 IN IP4 192.168.16.12 s=- c=IN IP4 192.168.16.12 t=0 0 m=audio 29000 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:40 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 [8] 2007/06/01 18:20:24: Resolve destination 43549: url sip:0664771607997@192.168.16.12 [8] 2007/06/01 18:20:24: Resolve destination 43549: udp 192.168.16.12 5060 [8] 2007/06/01 18:20:24: Send Packet ACK [7] 2007/06/01 18:20:24: SIP Tx udp:192.168.16.12:5060: ACK sip:0664771607997@192.168.16.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-39707571dca058a987d135230482d4e8;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 Call-ID: 841706e2@pbx CSeq: 17112 ACK Max-Forwards: 70 Contact: <sip:sippbx@192.168.16.110:5060;transport=udp> Remote-Party-ID: "sippbx" <sip:sippbx@192.168.16.12>;party=calling;screen=yes Content-Length: 0 [8] 2007/06/01 18:20:24: Resolve destination 43550: a udp 192.168.16.124 5060 [8] 2007/06/01 18:20:24: Resolve destination 43550: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:24: Send Packet 200 [7] 2007/06/01 18:20:24: SIP Tx udp:192.168.16.124:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK421753744;rport=5060 From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 INVITE Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 152 v=0 o=- 29725 29725 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=audio 49966 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=ptime:20 a=sendrecv [7] 2007/06/01 18:20:24: SIP Rx udp:192.168.16.124:5060: ACK sip:15@192.168.16.110:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK662941648;rport From: "Fax" <sip:15@192.168.16.110>;tag=1026814437 To: <sip:0664771607997@192.168.16.110>;tag=ae3503080a Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 61 ACK Contact: <sip:15@192.168.16.124:5060> Max-Forwards: 70 Supported: replaces, path User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 [7] 2007/06/01 18:20:24: SIP Rx udp:192.168.16.124:5060: REGISTER sip:192.168.16.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK529997551;rport Route: <sip:192.168.16.110:5060;lr> From: <sip:15@192.168.16.110>;tag=780704202 To: <sip:15@192.168.16.110> Call-ID: 1506464803-5060-1@192.168.16.124 CSeq: 2020 REGISTER Contact: <sip:15@192.168.16.124:5060> Authorization: Digest username="15", realm="192.168.16.110", nonce="2fa227c4316caf5070e594d543c13f78", uri="sip:192.168.16.110", response="98e3da31d23059ab4a7152113633198d", algorithm=MD5 Expires: 3600 Max-Forwards: 70 User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 [8] 2007/06/01 18:20:24: Resolve destination 43551: a udp 192.168.16.124 5060 [8] 2007/06/01 18:20:24: Resolve destination 43551: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:24: Send Packet 200 [7] 2007/06/01 18:20:24: SIP Tx udp:192.168.16.124:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK529997551;rport=5060 From: <sip:15@192.168.16.110>;tag=780704202 To: <sip:15@192.168.16.110>;tag=790fb62a3d Call-ID: 1506464803-5060-1@192.168.16.124 CSeq: 2020 REGISTER Contact: <sip:15@192.168.16.124:5060>;expires=60 Content-Length: 0 [7] 2007/06/01 18:20:31: SIP Rx udp:192.168.16.12:5060: INVITE sip:sippbx@192.168.16.110:5060;transport=udp SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP 192.168.16.12:5060;rport;branch=z9hG4bK04351950256772412077751 From: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 To: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 Contact: <sip:0664771607997@192.168.16.12> User-Agent: TELES.VoIPBOX 13.0b 885 Call-ID: 841706e2@pbx CSeq: 17113 INVITE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Content-Type: application/sdp Content-Length: 290 v=0 o=- 277 1 IN IP4 192.168.16.12 s=- c=IN IP4 192.168.16.12 t=0 0 m=image 29000 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 [8] 2007/06/01 18:20:31: Resolve destination 43555: a udp 192.168.16.12 5060 [8] 2007/06/01 18:20:31: Resolve destination 43555: udp 192.168.16.12 5060 [8] 2007/06/01 18:20:31: Send Packet 100 [7] 2007/06/01 18:20:31: SIP Tx udp:192.168.16.12:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.12:5060;rport=5060;branch=z9hG4bK04351950256772412077751 From: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 To: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 Call-ID: 841706e2@pbx CSeq: 17113 INVITE Content-Length: 0 [7] 2007/06/01 18:20:31: UDP: Opening socket on port 52350 [7] 2007/06/01 18:20:31: UDP: Opening socket on port 52351 [7] 2007/06/01 18:20:31: UDP: Opening socket on port 52670 [7] 2007/06/01 18:20:31: UDP: Opening socket on port 52671 [8] 2007/06/01 18:20:31: Resolve destination 43556: url sip:15@192.168.16.124:5060 [8] 2007/06/01 18:20:31: Resolve destination 43556: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:31: Send Packet INVITE [7] 2007/06/01 18:20:31: SIP Tx udp:192.168.16.124:5060: INVITE sip:15@192.168.16.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-a65ceff6d69e0acae419decacfc1a337;rport From: <sip:0664771607997@192.168.16.110>;tag=ae3503080a To: "Fax" <sip:15@192.168.16.110>;tag=1026814437 Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 27896 INVITE Max-Forwards: 70 Contact: <sip:15@192.168.16.110:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 298 v=0 o=- 29725 29727 IN IP4 192.168.16.110 s=- c=IN IP4 192.168.16.110 t=0 0 m=image 52670 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 [5] 2007/06/01 18:20:31: UDP: recvfrom returns WSAEWOULDBLOCK [7] 2007/06/01 18:20:31: SIP Rx udp:192.168.16.124:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-a65ceff6d69e0acae419decacfc1a337;rport=5060 From: <sip:0664771607997@192.168.16.110>;tag=ae3503080a To: "Fax" <sip:15@192.168.16.110>;tag=1026814437 Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 27896 INVITE Contact: <sip:15@192.168.16.124:5060> Supported: replaces, path User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 [7] 2007/06/01 18:20:31: SIP Rx udp:192.168.16.124:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-a65ceff6d69e0acae419decacfc1a337;rport=5060 From: <sip:0664771607997@192.168.16.110>;tag=ae3503080a To: "Fax" <sip:15@192.168.16.110>;tag=1026814437 Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 27896 INVITE Contact: <sip:15@192.168.16.124:5060> Supported: replaces, path User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Content-Length: 351 v=0 o=15 8000 8001 IN IP4 192.168.16.124 s=SIP Call c=IN IP4 192.168.16.124 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy [8] 2007/06/01 18:20:31: Resolve destination 43557: url sip:15@192.168.16.124:5060 [8] 2007/06/01 18:20:31: Resolve destination 43557: udp 192.168.16.124 5060 [8] 2007/06/01 18:20:31: Send Packet ACK [7] 2007/06/01 18:20:31: SIP Tx udp:192.168.16.124:5060: ACK sip:15@192.168.16.124:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-8c2bf4c60797fca57c45945d91d80936;rport From: <sip:0664771607997@192.168.16.110>;tag=ae3503080a To: "Fax" <sip:15@192.168.16.110>;tag=1026814437 Call-ID: 167922533-5060-7@192.168.16.124 CSeq: 27896 ACK Max-Forwards: 70 Contact: <sip:15@192.168.16.110:5060> Content-Length: 0 [8] 2007/06/01 18:20:31: Resolve destination 43558: a udp 192.168.16.12 5060 [8] 2007/06/01 18:20:31: Resolve destination 43558: udp 192.168.16.12 5060 [8] 2007/06/01 18:20:31: Send Packet 200 [7] 2007/06/01 18:20:31: SIP Tx udp:192.168.16.12:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.16.12:5060;rport=5060;branch=z9hG4bK04351950256772412077751 From: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 To: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 Call-ID: 841706e2@pbx CSeq: 17113 INVITE Contact: <sip:sippbx@192.168.16.110:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE Accept: application/sdp User-Agent: FL-Connect IP-PBX/2.0.3.1715 Content-Type: application/sdp Content-Length: 353 v=0 o=- 11568 11570 IN IP4 192.168.16.110 s=SIP Call c=IN IP4 192.168.16.110 t=0 0 m=image 52350 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy [7] 2007/06/01 18:20:31: SIP Rx udp:192.168.16.12:5060: ACK sip:sippbx@192.168.16.110:5060;transport=udp SIP/2.0 Max-Forwards: 50 Via: SIP/2.0/UDP 192.168.16.12:5060;rport;branch=z9hG4bK16087354823085853174033 From: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 To: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 Contact: <sip:0664771607997@192.168.16.12> User-Agent: TELES.VoIPBOX 13.0b 885 Call-ID: 841706e2@pbx CSeq: 17113 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Content-Length: 0 ... and now the malformed packets sending beginns and the fax does not hear anything anymore.... [7] 2007/06/01 18:21:04: SIP Rx udp:192.168.16.124:5060: REGISTER sip:192.168.16.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK1124454462;rport Route: <sip:192.168.16.110:5060;lr> From: <sip:15@192.168.16.110>;tag=780704202 To: <sip:15@192.168.16.110> Call-ID: 1506464803-5060-1@192.168.16.124 CSeq: 2021 REGISTER Contact: <sip:15@192.168.16.124:5060> Authorization: Digest username="15", realm="192.168.16.110", nonce="2fa227c4316caf5070e594d543c13f78", uri="sip:192.168.16.110", response="98e3da31d23059ab4a7152113633198d", algorithm=MD5 Expires: 3600 Max-Forwards: 70 User-Agent: Grandstream HT-502 1.0.0.39 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 [8] 2007/06/01 18:21:04: Resolve destination 43575: a udp 192.168.16.124 5060 [8] 2007/06/01 18:21:04: Resolve destination 43575: udp 192.168.16.124 5060 [8] 2007/06/01 18:21:04: Send Packet 200 [7] 2007/06/01 18:21:04: SIP Tx udp:192.168.16.124:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.16.124:5060;branch=z9hG4bK1124454462;rport=5060 From: <sip:15@192.168.16.110>;tag=780704202 To: <sip:15@192.168.16.110>;tag=790fb62a3d Call-ID: 1506464803-5060-1@192.168.16.124 CSeq: 2021 REGISTER Contact: <sip:15@192.168.16.124:5060>;expires=60 Content-Length: 0 [7] 2007/06/01 18:21:07: Other Ports: 1 [7] 2007/06/01 18:21:07: Call Port: 841706e2@pbx#23234 [8] 2007/06/01 18:21:07: Resolve destination 43579: url sip:0664771607997@192.168.16.12 [8] 2007/06/01 18:21:07: Resolve destination 43579: udp 192.168.16.12 5060 [8] 2007/06/01 18:21:07: Send Packet BYE [7] 2007/06/01 18:21:07: SIP Tx udp:192.168.16.12:5060: BYE sip:0664771607997@192.168.16.12 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-3648224c7aac549c0b09cfdd2d85cbf5;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 Call-ID: 841706e2@pbx CSeq: 17113 BYE Max-Forwards: 70 Contact: <sip:sippbx@192.168.16.110:5060;transport=udp> RTP-RxStat: Dur=51,Pkt=1270,Oct=421640,Underun=494 RTP-TxStat: Dur=43,Pkt=93,Oct=29604 Remote-Party-ID: "sippbx" <sip:sippbx@192.168.16.12>;party=calling;screen=yes Content-Length: 0 [8] 2007/06/01 18:21:07: Resolve destination 43580: url sip:35@192.168.16.123:2054;line=4ym9kfvn [8] 2007/06/01 18:21:07: Resolve destination 43580: udp 192.168.16.123 2054 [8] 2007/06/01 18:21:07: Send Packet NOTIFY [7] 2007/06/01 18:21:07: SIP Tx udp:192.168.16.123:2054: NOTIFY sip:35@192.168.16.123:2054;line=4ym9kfvn SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-888c61cd7c75989ae6ad5634a91b8444;rport From: <sip:15@localhost;user=phone>;tag=003530fce3 To: <sip:35@localhost>;tag=qdifvavd0l Call-ID: 3c26700e3b25-np70hd4mtm3d CSeq: 1373 NOTIFY Max-Forwards: 70 Contact: <sip:192.168.16.110:5060;transport=udp> Event: dialog Subscription-State: active;expires=52 Content-Type: application/dialog-info+xml Content-Length: 147 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="51" state="full" entity="sip:15@localhost"></dialog-info> [8] 2007/06/01 18:21:07: Resolve destination 43581: url sip:25@192.168.16.121:1030;line=ymmilo56 [8] 2007/06/01 18:21:07: Resolve destination 43581: udp 192.168.16.121 1030 [8] 2007/06/01 18:21:07: Send Packet NOTIFY [7] 2007/06/01 18:21:07: SIP Tx udp:192.168.16.121:1030: NOTIFY sip:25@192.168.16.121:1030;line=ymmilo56 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-30132f7a1d1f797d683a25286b1f21ac;rport From: <sip:15@localhost>;tag=6029d08633 To: <sip:25@localhost>;tag=gjc1casrya Call-ID: 3c26701bc9ca-8rwmzrkyi31t CSeq: 24243 NOTIFY Max-Forwards: 70 Contact: <sip:192.168.16.110:5060;transport=udp> Event: dialog Subscription-State: active;expires=69 Content-Type: application/dialog-info+xml Content-Length: 147 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="51" state="full" entity="sip:15@localhost"></dialog-info> [7] 2007/06/01 18:21:07: SIP Rx udp:192.168.16.12:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-3648224c7aac549c0b09cfdd2d85cbf5;rport From: "Fax " <sip:+43724235108815@192.168.16.12>;tag=23234 To: <sip:0664771607997@192.168.16.12;user=phone>;tag=637880302461171716578104820567 Call-ID: 841706e2@pbx CSeq: 17113 BYE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER Contact: <sip:0664771607997@192.168.16.12> User-Agent: TELES.VoIPBOX 13.0b 885 Content-Length: 0 [5] 2007/06/01 18:21:07: BYE Response: Terminate 841706e2@pbx [7] 2007/06/01 18:21:07: SIP Rx udp:192.168.16.121:1030: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-30132f7a1d1f797d683a25286b1f21ac;rport=5060 f: <sip:15@localhost>;tag=6029d08633 t: <sip:25@localhost>;tag=gjc1casrya i: 3c26701bc9ca-8rwmzrkyi31t CSeq: 24243 NOTIFY l: 0 [7] 2007/06/01 18:21:07: SIP Rx udp:192.168.16.123:2054: SIP/2.0 200 Ok v: SIP/2.0/UDP 192.168.16.110:5060;branch=z9hG4bK-888c61cd7c75989ae6ad5634a91b8444;rport=5060 f: <sip:15@localhost;user=phone>;tag=003530fce3 t: <sip:35@localhost>;tag=qdifvavd0l i: 3c26700e3b25-np70hd4mtm3d CSeq: 1373 NOTIFY l: 0

Hi! We have managed to set up T38 on both our, our gateway to the outside world and our ATA. If the ATA registers on the Gateway, T38 works fine. However, if i use the ATA as a PBXnSIP extension, and the gateway sends a re-invite, i receive the sip message "NOT SUPPORTED" from the PBX. Now, is the PBX T38 compatible or not???? And if yes, why does it react this way?

By buying additional codec licenses. Please contact your reseller.

We are selling and installing the 370 as well. Unfortunately PnP does not work with PBXnSIP. The main key for the phone to work is the correct IDENTITY setting. Make sure it is exactly the same setting as one of your 300, 320 or 360 phones which have been configured with PnP (except login data and passowrd of course). Then you need to specify the starcodes in the advanced settings. If you do that, it?ll work fine!

TLS switched off... Thanks for the hint. One more question.... We are working with Config files inside the HTML directory for the snom phones. Is there any describtion of this file available? Is it possible to perpare a file for each snom phone? This would help with big PBXs.

We also tested several combinations. Faxing over Ethernet, so FAX OVER VOIP no matter if LAN or WAN does not work most of the time. We now use a T38 ISDN Gateway (TELES VoipGate) together with a Patton/Inalp ATA. Both talk T.38. That works fine with all faxes.

We have serious problems with SNOM sending TLS messages to PBXnSIP. We do not have a certificate in our pbx as we run it on windows2000 embedded and we can not generate a CSR. After a couple of hours, the phones do not register anymore and i need to reboot them. According to SNOM, TLS can not be deactivated. I do not believe that. Does anyone know how to deactivate TLS on SNOM?

Yes... sorted ! Thanks!

Hi! I have a very similar problem. Our VOIP Gateway to the outside world supports T38. The problem is, i simply cant get the SPA to work with t38. I made traces which very clearly show that the SPA does not switch to T38. Even when the re-invite is sent from our gateway. It sticks to g711. I will now try a Patton/Inalp ATA and the Grandstream Adapter which seems to work better. I shall keep you posted.

Sorry, did i understand that correctly? PBXnSIP + SNOM works fine?? Let me tell you what we try to do..... We try to get SNOM and PBXnSIP WORK.... And... here is our list of problems. 1. SNOM TLS ... Is not supported with PBXnSIP. Phones no moe register after a couple of hours. Need to be restarted. 2. SNOM SRTP ... After 7-10 mins, there is white noise on the line. SRTP with snom does not work. 3. DND function ... If you switch DND on and off a couple of times in a short period of time, the phone does not register anymore 4. SNOM 370 ... not supported for PNP 5. Installation with PNP does not work at all. Several phones register with the same extension, no matter if the MAC address is set or not. etc.... It does not work at all!!

Hello! SNOM and PBXnSIP seem to have a compatibility issue. Here is what we tested: Short calls (less than 7 mins) allways work fine and the quality is good. After approx. seven minutes (no matter if calling to external number or receiving a call from a external or internal call extension to extension), audio gets lost and there is only noise on one leg. We made a ethereal trace and could see that the connection from the PBX to the phone got lost. "Destination unreachable, Port unreachable". The problem does not occur with Linksys or Polycom phones. Our setup is: PBXNSIP 2.0.3.1715 (Win32) on Windows 2000SP4. SNOM 360 and 320 version 6.5.8 Windows 2003R2 Server with DHCP activated TELES VoIPBOX ISDN v13.0 All Connected to a CISCO Catalyst 2950-24 IOS 12.1 A trouble ticket wit SNOM has been opened. I am wondering you can reproduce the problem?

In Austria and Germany, all you get from the Telco is a ISDN Telephone number. It is up to you if you use it for a PBX or not. We, for example have +437242351088. If a customer dials this number he reaches our PBX. The PBX waits for 3 seconds if a extension number is send or not. This is done by overlap dialing in the ISDN B-Channel. If a extension is dialed, the PBX routes to the extension. If nothing is dialed, or the extension is unknown, it connects to the operator or whatever. Thats how it works here. I have no influence in that but maybe i can change the incommig number in the gateway. I need to check. , Thanks anyways.

Oh... then welcome to Europe! The Gateway only sends the extension number. Nothing else. And if there is no extension number it is annonymous. Thats normal here. BTW, your pattern does not work.

Does not work.... Here is a trace from our VOIP-ISDN Gateway Can you see the "To: <sip:192.168.16.11>" ? There is no extension! [52:49.90] i[04]: pstnrcv setup dad DF: oad 06641607997 cc 40 id 44d027 c/c=4/1 [52:49.92] i[04]: pstnrcv get_voipcfg <DF> compr <A> [52:49.93] x[04]: sip_send 754 192.168.16.11:5060 [52:49.95] x[04]: INVITE sip:192.168.16.11 SIP/2.0 [52:49.96] x[04]: Max-Forwards: 50 [52:49.98] x[04]: Via: SIP/2.0/UDP 192.168.16.12:5060;rport;branch=z9hG4bK60536325956379204269682 [52:49.99] x[04]: From: <sip:06641607997@192.168.16.12>;tag=676933635591324563097352172832 [52:50.01] x[04]: To: <sip:192.168.16.11> [52:50.03] x[04]: Call-ID: 856397878008217682549955350971@192.168.16.12 [52:50.04] x[04]: CSeq: 1 INVITE [52:50.06] x[04]: P-Asserted-Identity: tel:06641607997 [52:50.07] x[04]: Contact: <sip:06641607997@192.168.16.12> [52:50.09] x[04]: User-Agent: TELES.VoIPBOX 13.0b 885 [52:50.10] x[04]: Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER [52:50.12] x[04]: Timestamp: 1178524369 [52:50.14] x[04]: Content-Type: application/sdp [52:50.15] x[04]: Content-Length: 187 [52:50.17] x[04]: v=0 [52:50.18] x[04]: o=- 39 1 IN IP4 192.168.16.12 [52:50.20] x[04]: s=- [52:50.21] x[04]: c=IN IP4 192.168.16.12 [52:50.23] x[04]: t=0 0 [52:50.24] x[04]: m=audio 29000 RTP/AVP 8 101 [52:50.26] x[04]: a=rtpmap:8 PCMA/8000 [52:50.28] x[04]: a=ptime:20 [52:50.29] x[04]: a=rtpmap:101 telephone-event/8000 [52:50.31] x[04]: a=fmtp:101 0-15 [52:50.32] x[04]: sipsnd invite dad cr 27 to 192.168.16.11 rc 754 [52:50.34] y[04]: sip_recvfrom 312 192.168.16.11:5060 [52:50.35] y[04]: SIP/2.0 100 Trying [52:50.37] y[04]: Via: SIP/2.0/UDP 192.168.16.12:5060;rport=5060;branch=z9hG4bK60536325956379204269682 [52:50.37] y[04]: From: <sip:06641607997@192.168.16.12>;tag=676933635591324563097352172832 [52:50.40] y[04]: To: <sip:192.168.16.11>;tag=bf22502ab2 [52:50.42] y[04]: Call-ID: 856397878008217682549955350971@192.168.16.12 [52:50.43] y[04]: CSeq: 1 INVITE [52:50.45] y[04]: Content-Length: 0 [52:50.46] y[04]: siprcv trying from 192.168.16.11 cr 27 [52:50.48] y[04]: sip_recvfrom 599 192.168.16.11:5060 [52:50.49] y[04]: SIP/2.0 404 Not Found [52:50.51] y[04]: Via: SIP/2.0/UDP 192.168.16.12:5060;rport=5060;branch=z9hG4bK60536325956379204269682 [52:50.53] y[04]: From: <sip:06641607997@192.168.16.12>;tag=676933635591324563097352172832 [52:50.54] y[04]: To: <sip:192.168.16.11>;tag=bf22502ab2 [52:50.56] y[04]: Call-ID: 856397878008217682549955350971@192.168.16.12 [52:50.57] y[04]: CSeq: 1 INVITE [52:50.59] y[04]: Contact: <sip:sippbx@192.168.16.11:5060;transport=udp> [52:50.60] y[04]: Supported: 100rel, replaces, norefersub [52:50.62] y[04]: Allow-Events: refer [52:50.64] y[04]: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, PRACK, INFO, PUBLISH, NOTIFY, SUBSCRIBE, MESSAGE [52:50.65] y[04]: Accept: application/sdp [52:50.67] y[04]: User-Agent: FL-Connect IP-PBX/2.0.3.1715 [52:50.68] y[04]: Content-Length: 0 [52:50.70] y[04]: siprcv terminate (404) from 192.168.16.11 cr 27 [52:50.71] x[04]: sip_send 513 192.168.16.11:5060 [52:50.73] x[04]: ACK sip:sippbx@192.168.16.11:5060;transport=udp SIP/2.0 [52:50.74] x[04]: Max-Forwards: 50 [52:50.76] x[04]: Via: SIP/2.0/UDP 192.168.16.12:5060;rport=5060;branch=z9hG4bK60536325956379204269682 [52:50.78] x[04]: From: <sip:06641607997@192.168.16.12>;tag=676933635591324563097352172832 [52:50.79] x[04]: To: <sip:192.168.16.11>;tag=bf22502ab2 [52:50.81] x[04]: Contact: <sip:06641607997@192.168.16.12> [52:50.82] x[04]: User-Agent: TELES.VoIPBOX 13.0b 885 [52:50.84] x[04]: Call-ID: 856397878008217682549955350971@192.168.16.12 [52:50.85] x[04]: CSeq: 1 ACK [52:50.87] x[04]: Allow: INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER,PRACK,INFO,NOTIFY,REFER [52:50.89] x[04]: Content-Length: 0 [52:50.90] x[04]: sipsnd ack cr 27 to 192.168.16.11 rc 513

I am trying to do the following: Currently we are connected to 2 telcos of which one hosts our tel. number. We want to receive calls thru telco1 and send calls to both depending on the prefix. Incomming calls (Telco1): Extension dialed -> route to the selected extension number. No extension or unknown extension -> route to extension 25 Outgoing calls (Telco2): Everything comming from the PBX should be routed to Telco2 except calls from Extension 15 (fax) which should be routed thru telco1. (Telco2 is not able to switch to T38). Prefix of a external line should be 0. How can i do that? Thanks in advance!

Is there a chart available which shows the recomended CPU power in relation to concurrent alls based on the windows pbxnsip built on a win2000 system? I have seen your suggestions in the wiki but i am not sure how much cpu power the os itself needs. Could anyone help me with that?

Hello! We understood that the PBX is mainly controlled by the Dialing Plan. However, we find it pretty complicated to generate a propper dialingplan as noone here is familiar with regular expressions. Is there a kind of Dialing plan generator available or do you plan to implement such a tool in future releases? Thanks!