p800aul

Was this ever done, if so could you point me to it. If not what's the best workround? thanks Regards Paul

Here's my running config you will need to edit it to work with your setup, unless the snom one pbx is at 192.168.1.13 on your network. edit at interface sip IF_SIP_1 and interface sip IF_SIP_2. As far as step by step you load this file in to the patton (after the edit) setup a trunk as per the previous post and it should work, as it does for me. If you can not get it to work and you are sure it's the patton give patton tech support a call they are very good indeed and can remotly set the thing up for you, if you are really stuck. Snom one should just work as setup above, if it doesn't publish your config and if i cannot see whats wrong someone else will (i'm new aswell) I assume you have a dialplan? regards Paul pattonrun.txt

You may want to check this post out http://forum.snom.com/index.php?showtopic=5804 Worth reading it all I now have it working by the way Regards Paul

Hi This is how our 4112 2 line fxo is setup on a trunk. Notes are (in brackets) I hope it helps if you need my running config of the patton let me know and i'll post it. Regards Paul # Trunk 5 in domain localhost Name: Patton Type: register To: sip (nothing set) RegPass: ******** (nothing set) Direction: (Inbound outbound) Disabled: false Global: false Display: RegAccount: RegRegistrar: 192.168.1.XXX (the IP address of the Patton) RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.1.XXX (the IP address of the Patton) Ani: DialExtension: 72 (the Hunt group) Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: only_5xx Privacy: false Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: Colines: DialogPermission:

Config, i was told originally that not having a line attached to the second FXO port wouldn't matter, it seems it does! They made a couple of changes to my original config the settings on the FXO interface so it will recognize a longer or shorter tone break and see this as a disconnect and go back on-hook to be ready to accept another call. Secondly, disabled cyclic routing in your hunt group. Since I only have one interface working at the moment, the hunt will now try the first interface over and over. Apparently this is easy change back when I need that 2nd interface in routing. Out of interest for everyone, here is my running config for a Patton 4112 with only one pstn line attached, I'm in the UK. Regards Paul #----------------------------------------------------------------# # # # SN4112/JO/EUI # # R5.2 2009-01-14 H323 SIP FXS FXO # # 2011-01-26T07:26:33 # # SN/00A0BA0609C0 # # Generated configuration file # # # #----------------------------------------------------------------# cli version 3.20 webserver port 80 language en sntp-client sntp-client server primary 194.35.252.7 port 123 version 4 sntp-client server secondary 194.164.127.5 port 123 version 4 sntp-client local-clock-offset system ic voice 0 low-bitrate-codec g729 profile ppp default profile call-progress-tone defaultDialtone play 1 1000 450 -6 profile call-progress-tone defaultAlertingtone play 1 1000 450 -13 pause 2 5000 profile call-progress-tone defaultBusytone play 1 300 450 -7 pause 2 300 profile call-progress-tone defaultReleasetone play 1 300 450 -7 pause 2 300 profile call-progress-tone defaultCongestiontone play 1 300 450 -7 pause 2 300 profile tone-set default profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 fax transmission 1 relay t38-udp fax transmission 2 bypass g711alaw64k profile pstn default profile sip default profile aaa default method 1 local method 2 none context ip router interface IF_IP_LAN ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface IF_IP_WAN ipaddress dhcp tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 192.168.1.1 0 context cs switch digit-collection timeout 2 interface sip IF_SIP_1 bind context sip-gateway GW_SIP_ALL_LINES route call dest-service HUNT_FXO remote 192.168.1.13 5060 early-connect early-disconnect address-translation outgoing-call request-uri user-part fix 10015 host-part to-header target-param none interface sip IF_SIP_2 bind context sip-gateway GW_SIP_ALL_LINES route call dest-service HUNT_FXO remote 192.168.1.13 5060 early-connect early-disconnect address-translation outgoing-call request-uri user-part fix 10016 host-part to-header target-param none interface fxo IF_FXO_1 route call dest-interface IF_SIP_1 loop-break-duration min 60 max 5000 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id dial-after timeout 1 mute-dialing interface fxo IF_FXO_2 route call dest-interface IF_SIP_2 loop-break-duration min 100 max 500 disconnect-signal loop-break disconnect-signal busy-tone ring-number on-caller-id dial-after timeout 1 mute-dialing service hunt-group HUNT_FXO drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_FXO_1 route call 2 dest-interface IF_FXO_2 context cs switch no shutdown authentication-service AS_ALL_LINES username 10015 password Z+ApY8PXmFjMRxFr04ls2w== encrypted username 10016 password c7k7vrPq2MMY+mdxPJS6aQ== encrypted location-service LS_ALL_LINES identity 10015 identity 10016 context sip-gateway GW_SIP_ALL_LINES interface LAN bind interface IF_IP_LAN context router port 5060 context sip-gateway GW_SIP_ALL_LINES no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface IF_IP_LAN router no shutdown port fxo 0 0 use profile fxo gb encapsulation cc-fxo bind interface IF_FXO_1 switch no shutdown port fxo 0 1 use profile fxo gb encapsulation cc-fxo bind interface IF_FXO_2 switch shutdown

here you go by the way Patton fixed it Microsoft Windows XP [Version 5.1.2600] © Copyright 1985-2001 Microsoft Corp. C:\Documents and Settings\paul>route print =========================================================================== Interface List 0x1 ........................... MS TCP Loopback interface 0x2 ...00 24 1d a0 7f 65 ...... Realtek PCIe FE Family Controller - Packet Sched uler Miniport =========================================================================== =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 192.168.1.1 192.168.1.13 20 127.0.0.0 255.0.0.0 127.0.0.1 127.0.0.1 1 192.168.1.0 255.255.255.0 192.168.1.13 192.168.1.13 20 192.168.1.13 255.255.255.255 127.0.0.1 127.0.0.1 20 192.168.1.255 255.255.255.255 192.168.1.13 192.168.1.13 20 224.0.0.0 240.0.0.0 192.168.1.13 192.168.1.13 20 255.255.255.255 255.255.255.255 192.168.1.13 192.168.1.13 1 Default Gateway: 192.168.1.1 =========================================================================== Persistent Routes: None Regards Paul

So is the routing in Matt's post ok for me?

Hi pbxnsip Thanks for your reply. I've been back on to patton as the issue seems to be a 502 from that trunk I've run a debug on the patton and this hopefully will give them a clue as to why this is happening. It only happens on alternate calls i.e. a call to 123321 goes through, hang up, call to 123321 busy tone (502), hang up, call to 123321 goes through, this is regardless of time between the calls. The rest of the system works i have the pbx on the dmz and as it's a simple system i set it up using this from Matt post xx.xx.xx.xx is the public when i get a solution or not i'll come back Regards Paul

Hi Any help with this would be great. I have a patton 4112 2 x fxo gateway on Snom One. When making a call via this gateway the call will either connect or give a busy tone alternately, this behaviour is consistent, i.e. call to 01246123123 call rings and works fine, hang up, call 01246123123 line busy, hang up, call 01246123123 call rings and works fine and so on..... I have had the config of the patton checked by patton and they don’t see any issues which could cause this behaviour, we tried changing a few things on the patton with no effect. The logs etc are below along with the trunk set up. Thanks for any help regards Paul the trunk is setup: # Trunk 5 in domain localhost Name: Patton Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: 192.168.1.200 RegKeep: RegUser: Icid: Require: OutboundProxy: 192.168.1.200 Ani: DialExtension: 44 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: only_5xx Privacy: pai Glob: RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: Colines: 2 DialogPermission: Log from a succesful call [9] 2011/01/23 22:32:47: [7] 2011/01/23 22:32:50: INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: m6j5w1sdsk CSeq: 22153 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Content-Type: application/sdp Content-Length: 398 v=0 o=root 1565728340 1565728341 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 52578 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:VUyNQYRiVjnhewe7vV1qF+eJ9VtmWTiW1RZOyQT6|2^31 a=sendrecv [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22153 INVITE Content-Length: 0 [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22153 INVITE User-Agent: snom-PBX/4.2.0.3974 WWW-Authenticate: Digest realm="192.168.1.13",nonce="380850f72b1790232735c52375b1d44a",domain="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Length: 0 [7] 2011/01/23 22:32:50: ACK sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-kda1ay;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22153 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [7] 2011/01/23 22:32:50: INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Authorization: Digest realm="192.168.1.13",nonce="380850f72b1790232735c52375b1d44a",response="6929aee06e8ccb51bbe0ab176106929d",username="45",uri="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 398 v=0 o=root 1565728340 1565728341 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 52578 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:VUyNQYRiVjnhewe7vV1qF+eJ9VtmWTiW1RZOyQT6|2^31 a=sendrecv [8] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Content-Length: 0 [8] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: INVITE sip:263016@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone> Call-ID: f5058ca5@pbx CSeq: 8184 INVITE Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Type: application/sdp Content-Length: 327 v=0 o=- 58077 58077 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 63112 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=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 324 v=0 o=- 43827 43827 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 54492 RTP/AVP 0 8 9 2 3 96 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:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2011/01/23 22:32:50: PRACK sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-0tvq9n;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22155 PRACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" RAck: 1 22154 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Content-Length: 0 [9] 2011/01/23 22:32:50: [7] 2011/01/23 22:32:50: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-0tvq9n;rport=4043;received=192.168.1.1 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22155 PRACK Contact: <sip:45@xx.xx.xx.xx:5060> User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [7] 2011/01/23 22:32:51: SIP/2.0 100 Trying Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport=5060;received=192.168.1.13 From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone> Call-ID: f5058ca5@pbx CSeq: 8184 INVITE Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26 Content-Length: 0 [9] 2011/01/23 22:32:51: [7] 2011/01/23 22:32:54: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-a886bdd55a936239c274a5b1624fd6bb;rport=5060;received=192.168.1.13 From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773 Call-ID: f5058ca5@pbx CSeq: 8184 INVITE Contact: <sip:263016@192.168.1.200:5060> Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26 Supported: replaces Content-Type: application/sdp Content-Length: 221 v=0 o=MxSIP 0 57 IN IP4 192.168.1.200 s=SIP Call c=IN IP4 192.168.1.200 t=0 0 m=audio 4976 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [7] 2011/01/23 22:32:54: ACK sip:263016@192.168.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-291e98728a7e6723bd58601166585878;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773 Call-ID: f5058ca5@pbx CSeq: 8184 ACK Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Length: 0 [9] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [9] 2011/01/23 22:32:54: [7] 2011/01/23 22:32:54: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-n5fh0q;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Content-Type: application/sdp Content-Length: 324 v=0 o=- 43827 43827 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 54492 RTP/AVP 0 8 9 2 3 96 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:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2011/01/23 22:32:54: ACK sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-zaqp37;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22154 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [8] 2011/01/23 22:32:59: [9] 2011/01/23 22:32:59: [7] 2011/01/23 22:33:00: BYE sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-ce9s8k;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22156 BYE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub Content-Length: 0 [9] 2011/01/23 22:33:00: [7] 2011/01/23 22:33:00: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-ce9s8k;rport=4043;received=192.168.1.1 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=ao053b To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=827a0a9196 Call-ID: m6j5w1sdsk CSeq: 22156 BYE Contact: <sip:45@xx.xx.xx.xx:5060> User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [9] 2011/01/23 22:33:00: [9] 2011/01/23 22:33:00: [7] 2011/01/23 22:33:00: BYE sip:263016@192.168.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-ac3b9d3338c98fa09f1c607eeeca5213;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=81 To: <sip:263016@192.168.1.200;user=phone>;tag=2546102773 Call-ID: f5058ca5@pbx CSeq: 8185 BYE Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Length: 0 And the log from a failed call made stright after the above. INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: o77i5idc96 CSeq: 12645 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Content-Type: application/sdp Content-Length: 398 v=0 o=root 1833475499 1833475500 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 56684 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZT6rmB8cjbvcKGdGKF5E1VXrx1ZBnP7/04nGSRA7|2^31 a=sendrecv [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:64380 [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:64381 [9] 2011/01/23 22:31:51: Resolve 7529: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7529: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7529: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12645 INVITE Content-Length: 0 [9] 2011/01/23 22:31:51: Resolve 7530: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7530: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7530: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12645 INVITE User-Agent: snom-PBX/4.2.0.3974 WWW-Authenticate: Digest realm="192.168.1.13",nonce="c9244e6dab13e1f55b84ee9830031c0f",domain="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Length: 0 [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.2:4043: ACK sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-mldxp1;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12645 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.2:4043: INVITE sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone> Call-ID: o77i5idc96 CSeq: 12646 INVITE Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Supported: 100rel, replaces, norefersub User-Agent: snom-m9/9.2.42-a Authorization: Digest realm="192.168.1.13",nonce="c9244e6dab13e1f55b84ee9830031c0f",response="70b900d2b3d9130bd5d678d3f7985945",username="45",uri="sip:263016@192.168.1.13;user=phone",algorithm=MD5 Content-Type: application/sdp Content-Length: 398 v=0 o=root 1833475499 1833475500 IN IP4 192.168.1.2 s=- c=IN IP4 192.168.1.2 t=0 0 m=audio 56684 RTP/AVP 0 8 18 3 9 2 10 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=rtcp-xr:a=rtcp-xr:pkt-loss-rle pkt-dup-rle pkt-rcpt-times rcvr-rtt=sender stat-summary=loss,dup,jitt voip-metrics a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ZT6rmB8cjbvcKGdGKF5E1VXrx1ZBnP7/04nGSRA7|2^31 a=sendrecv [8] 2011/01/23 22:31:51: Tagging request with existing tag [9] 2011/01/23 22:31:51: Resolve 7531: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7531: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7531: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 INVITE Content-Length: 0 [8] 2011/01/23 22:31:51: Set the To domain based on From user 45@localhost [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:53478 [9] 2011/01/23 22:31:51: UDP: Opening socket on 0.0.0.0:53479 [9] 2011/01/23 22:31:51: Resolve 7532: url sip:192.168.1.200 [9] 2011/01/23 22:31:51: Resolve 7532: udp 192.168.1.200 5060 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.200:5060: INVITE sip:263016@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580 To: <sip:263016@192.168.1.200;user=phone> Call-ID: ed6c53a8@pbx CSeq: 27475 INVITE Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Type: application/sdp Content-Length: 327 v=0 o=- 28432 28432 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 53478 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=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [9] 2011/01/23 22:31:51: Resolve 7533: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7533: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7533: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.2:4043: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 324 v=0 o=- 16374 16374 IN IP4 xx.xx.xx.xx s=- c=IN IP4 xx.xx.xx.xx t=0 0 m=audio 64380 RTP/AVP 0 8 9 2 3 96 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:96 telephone-event/8000 a=fmtp:96 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.1:4043: PRACK sip:45@xx.xx.xx.xx:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-4jmwo3;rport From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12647 PRACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" RAck: 1 12646 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Content-Length: 0 [9] 2011/01/23 22:31:51: Resolve 7534: udp 192.168.1.1 4043 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.1:4043: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-4jmwo3;rport=4043;received=192.168.1.1 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12647 PRACK Contact: <sip:45@xx.xx.xx.xx:5060> User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [7] 2011/01/23 22:31:51: SIP Rx udp:192.168.1.200:5060: SIP/2.0 502 Bad Gateway Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport=5060;received=192.168.1.13 From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580 To: <sip:263016@192.168.1.200;user=phone>;tag=3034763092 Call-ID: ed6c53a8@pbx CSeq: 27475 INVITE Server: Patton SN4112 JO EUI 00A0BA0609C0 R5.2 2009-01-14 H323 SIP FXS FXO M5T SIP Stack/4.0.26.26 Content-Length: 0 [7] 2011/01/23 22:31:51: Call ed6c53a8@pbx: Clear last INVITE [9] 2011/01/23 22:31:51: Resolve 7535: url sip:192.168.1.200 [9] 2011/01/23 22:31:51: Resolve 7535: udp 192.168.1.200 5060 [7] 2011/01/23 22:31:51: SIP Tx udp:192.168.1.200:5060: ACK sip:263016@192.168.1.200;user=phone SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK-3b1c8f1e5d06889edea220cbcb412ec2;rport From: "Forty Five" <sip:45@localhost;user=phone>;tag=34580 To: <sip:263016@192.168.1.200;user=phone>;tag=3034763092 Call-ID: ed6c53a8@pbx CSeq: 27475 ACK Max-Forwards: 70 Contact: <sip:45@xx.xx.xx.xx:5060;transport=udp> P-Asserted-Identity: "Forty Five" <sip:45@localhost;user=phone> Content-Length: 0 [5] 2011/01/23 22:31:51: INVITE Response 502 Bad Gateway: Terminate ed6c53a8@pbx [9] 2011/01/23 22:31:51: Resolve 7536: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7536: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:51: Resolve 7536: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:52: SIP Tx udp:192.168.1.2:4043: SIP/2.0 502 Bad Gateway Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport=4043 From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 INVITE Contact: <sip:45@xx.xx.xx.xx:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3974 Content-Length: 0 [8] 2011/01/23 22:31:52: Hangup: Call 101 not found [7] 2011/01/23 22:31:52: SIP Rx udp:192.168.1.2:4043: ACK sip:263016@192.168.1.13;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-iy0q3r;rport Route: <sip:192.168.1.13;lr> From: "Bridget Ingle" <sip:45@192.168.1.13>;tag=6327xx To: "263016" <sip:263016@192.168.1.13;user=phone>;tag=9e07ec4866 Call-ID: o77i5idc96 CSeq: 12646 ACK Max-Forwards: 70 Contact: <sip:45@192.168.1.2:4043;transport=udp;line=lp1fdp>;reg-id=1;+sip.instance="<urn:uuid:04992dbf-b24d-47e2-ae0b-0a139978e882>" Content-Length: 0 [7] 2011/01/23 22:31:57: SIP Rx udp:192.168.1.2:4043: REGISTER sip:192.168.1.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-90ywmz;rport From: "Paul Stead" <sip:44@192.168.1.13>;tag=s8ozvp To: "Paul Stead" <sip:44@192.168.1.13> Call-ID: ulydh2y8@snom CSeq: 3435 REGISTER Max-Forwards: 70 Contact: <sip:44@192.168.1.2:4043;transport=udp;line=j5sjvx>;reg-id=1;+sip.instance="<urn:uuid:249f54b0-67ba-445c-8433-55ee8f3a7b1a>" Supported: path, outbound, gruu User-Agent: snom-m9/9.2.42-a Authorization: Digest realm="192.168.1.13",nonce="10ceb016de3d4209ddadda412473a800",response="9afc717ab4f9d532a315f8378102e9f7",username="44",uri="sip:192.168.1.13",algorithm=MD5 Expires: 354 Content-Length: 0 [9] 2011/01/23 22:31:57: Resolve 7537: aaaa udp 192.168.1.2 4043 [9] 2011/01/23 22:31:57: Resolve 7537: a udp 192.168.1.2 4043 [9] 2011/01/23 22:31:57: Resolve 7537: udp 192.168.1.2 4043 [7] 2011/01/23 22:31:57: SIP Tx udp:192.168.1.2:4043: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.1.2:4043;branch=z9hG4bK-90ywmz;rport=4043 From: "Paul Stead" <sip:44@192.168.1.13>;tag=s8ozvp To: "Paul Stead" <sip:44@192.168.1.13>;tag=49ef1d8f34 Call-ID: ulydh2y8@snom CSeq: 3435 REGISTER Contact: <sip:44@192.168.1.2:4043;transport=udp;line=j5sjvx>;expires=352 Require: outbound Supported: path Content-Length: 0

The news is the Patton was de bricked and with the help here it's now working with the Snom one PBX. Much joy here Ta Regards Paul

Yep agree trying to find one any suggestions UK Derbyshire / South Yorkshire? Oh and by the way Snom say, when asked about professional partner help, come on the course and take the exam

Hi I'm sure you are right Sir, but brick like it is, although I am sure it will be up and running later today when support get around to it. The reason I bought it was that it was first rate. I understand that the Portech is not a Patton but at least it gives me (a newbie) a clue on how to setup trunk for it (I think) The 3cx to be honest looks much easier to set up, I’ve watched a the video which seems to be, set up a pstn gateway, tell it it’s a Patton, the 3cx gives you the txt file to upload to the Patton and away it goes. My point with Snom One is that there is little or no help apart from you guys on how to set these things up. That said I want to persevere with Snom as I have bought their phones, but I am finding this very hard work indeed. Sir, I am very grateful for all of your help so far, if I could have sight of any of the ‘book’ which you think could help me that would be most helpful. All I am trying at this stage is as follows Snom one PBX Viop Trunk (currently working in a basic form) 2 pstn lines via a Patton 4112 (not set up at all yet) 1 x Remote office (not working trying to use x-lite at the moment looking at putting the pbx dmz side, will want a hard phone at sometime, found your review of the 300 most helpful thanks) Snom m9 phones (working very well with the PBX and VIOP trunk) Clearly that will not be the end of it but the above would be a start Regards Paul

Thank the Lord And thanks for your help

Just so I'm clear the DMZ (public address) is at the PBX end only. The remote office extension (Xlite - Snom 300) can be behind a firewall with nats?

I'll give it a shot I assume unless i do this i will have issues even if i use a snom 300 type phone remotely? Thanks Paul

Agreed(My 8 year old thinks that I'm second only to Lee Westwood golfing wise so at least I'm good at something ) Looking at the PDF it seems that it would only take two minutes to represent this as a How To for any FXO,GSM type gateway. It could even have the new and correct name on (Snom rather than pbxnsip .) Anyway does the PDF represent a good 'how to' for getting a patton 4112 2 x fxo working on a snom one pbx? thanks for your interest in my issue regards Paul

OK I updated the PBX and it still doen't work. Lets try and start at the begining Which ports do I need to be open at the remote office router for this to work, I check these to start with and move on if needed? Thanks for your input so far Regards Paul

Hi This PDF seems like just the thing I've been looking for, one of the most frustrating thing about Snom One for a newbie like me (I believe I have a broad understanding of technology) is that the whole voip thing appears to be a black art. This seems to be true even when taking to the experts, I phoned Patton(USA 1 hour)yesterday regarding a locked up 4112 and following instructions from the tech guy, we bricked the unit (I await further instructions via email from them). The point of this rant is the question, are there any "how to's" anywhere like the PDF discussed here if so where are they, if not I think it may be a good idea to have some. Once I have my 4112 backup I think I can use this PDF to help me set it up on the Snom One unless someone out there (a black art master)knows different. Regards Paul

The two issues are different, aren’t they? I did try the M9 soft on a clean note book and had the same issues, that is the M9 software not very responsive and resource hungry. I don’t have the note book with me today so cannot try it again from this remote office. You will also notice I said that the M9 is taking 50% of the CPU at rest (idle) the X-Lite is 0% - 1% I would suggest that this is therefore nothing to do with the pc I’m working on. In addition it allows me to open the browser once to insert the initial details but once I select save that’s when it freezes. So I can’t even change the settings. As I am new to this stuff if someone has their settings for a M9 (or X-Lite) to a snom one remote pbx I will be delighted to try those inserting my servers ip address. If you are using another soft phone (I assume you may be as you’ve only tested the M9) tell me which one and I’ll try that. thanks for your help so far regards Paul

Thanks I'm not (as far as I can see)using a stun. I will try the update later to see if that resolves the issue. I've just tried the M9 soft again taking off the X-Lite and it reaffirmed my first experience of it (hence trying X-Lite). Once I had given the 1st identity a setup in the web interface, the web interface froze so did the M9. In addition, at rest (no calls) the process m9Soft2.exe takes 50% of the CPU. X-Lite takes either 0% or 1%. This seems to screw everything else including the browser. Go figure Unless you think I'm doing something wrong with the M9? Regards Paul

Hi I've managed to get the x-lite to register to the snom one pbx at a remote office, the X-Lite can make trunk calls and extensions at the remote office can call the X-Lite. Making a call to an extension at the remote office is where I’m stuck, upon dialling the extension, X-Lite says the call is connected without ringing. If I then hang up on the X-Lite, when i check the PBX call logs it says the call is still connected and requires me to disconnect it at the PBX. I am clearly missing something, any clues? Regards Paul

Hi I assume there is a way of pulling (i.e. using) the domain address book from a snom one pbx to the M9 dect phones. I can find an guidance on this can it be done? Regards Paul

Ok now working with more help from a tech support guy help. In addtion to the above the following ports need to be opened 49152-64512 UDP inbound and outbound. In addition there was a DNS issue, sipgate.co.uk was not resolving correctly on the computer snom ONE was located and after this was changed to the IP address for sipgate.co.uk it worked Thanks for all your help and hopfully the thread will help others Regards Paul

By that i guess you mean that i have to open ports on my router. I figured that as I didn't for 3cx so i didn't for Snom One, if i do which ones? An easy answer to my question would be a screen print of your trunk setup (as you are using SipGate) with your details obscured to protect your account so i can check mine are the same along with the ports you've opened on your router. Or a copy from your text based edit window Thanks for you input so far Regards Paul

Hi Sorry to be a pain testing Snom One and while i got 3cx to setup without difficulty i like some of the options available on Snom One so I am now trying this. Using the following information (not real of course) what goes where in the trunk setup screen I've tried every which way to get it to work and unless i am missing another and further configuration i can not get it to work. needless to say I've looked for a "how to" i don't see one. SIP account data This account data must be used to configure your SIP device. sipgate provides detailed information for customers to configure their SIP devices to function correctly. More information can be found here. SIP-ID: 12345678 SIP password: GFFDRGG Status: offline Nickname: No name was set Edit -------------------------------------------------------------------------------- SIP Server data This account data must be used to configure your SIP-device. sipgate provides detailed information for customers to configure their SIP devices to function correctly. More information can be found here. Registry: sipgate.co.uk Proxy: sipgate.co.uk STUN: stun.sipgate.net:10000 NTP: ntp.sipgate.net i assume the trunk is all i need to setup the extensions are working after about one and one half an hours. You would think that snom m9 and snom one would just work out of the box By the way they sipgate DON'T provide detailed information for customers to configure their snom one SIP devices to function correctly. thanks regards Paul