Digital Pimp

Hi Mr X Thanks for the reply, below is the extract from the systems log file of an attempted call in - it's passed through the correct mobile number that I was using to test, just the call doesnt actually go any where. BTW I made a few errors on the IP addresses in my first post, now updated. I'm presuming the 404 messages below are playing a part in it failing to work. [5] 2012/07/16 20:06:42: SIP Rx udp:192.168.0.250:5062: INVITE sip:2000@192.168.0.249:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport Route: <sip:192.168.0.249:5060;lr> From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997 To: <sip:2000@192.168.0.249:5060> Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA CSeq: 10 INVITE Contact: <sip:123@192.168.0.250:5062> Max-Forwards: 70 User-Agent: Grandstream HT-503 V1.1B 1.0.6.13 chip V2.2 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 388 v=0 o=123 8002 8000 IN IP4 192.168.0.250 s=SIP Call c=IN IP4 192.168.0.250 t=0 0 m=audio 5013 RTP/AVP 0 8 4 18 2 97 102 100 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=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:102 G729E/8000 a=rtpmap:100 AAL2-G726-16/8000 [5] 2012/07/16 20:06:42: SIP Tx udp:192.168.0.250:5062: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport=5062 From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997 To: <sip:2000@192.168.0.249:5060>;tag=f791a28ac7 Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA CSeq: 10 INVITE Content-Length: 0 [5] 2012/07/16 20:06:42: SIP Tx udp:192.168.0.250:5062: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport=5062 From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997 To: <sip:2000@192.168.0.249:5060>;tag=f791a28ac7 Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA CSeq: 10 INVITE Contact: <sip:2000@192.168.0.249:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [5] 2012/07/16 20:06:43: SIP Tr udp:192.168.0.250:5062: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.250:5062;branch=z9hG4bK1568978287;rport=5062 From: <sip:0787xxxxxxx@192.168.0.249>;tag=608729997 To: <sip:2000@192.168.0.249:5060>;tag=f791a28ac7 Call-ID: 1270917753-5062-1@BJC.BGI.A.CFA CSeq: 10 INVITE Contact: <sip:2000@192.168.0.249:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0

Hi Guys I'm looking for some advice as to how to get my Grandstream HT503 to work as a PSTN gateway with Snon One, I’ll confess before I start that I’m new to Snom One and still finding my feet but so far much better than 3CX for what I need. I understand from reading other posts that the HT503 is not the best device to be using as a gateway but it’s what I’ve got to hand to tinker with. If any of you have managed to actually get this device to work could you let me know the settings you have used for both the trunk setup and the HT503? So far my tinkering has yielded it picking incoming calls up after two rings but immediately dropping to a dial tone, I haven’t worked out if the call is actually making it to Snom One – settings used below HT503 – 192.168.0.250 Snom One – 192.168.0.249 HT503 FXO SIP Server: 192.168.0.249 User ID: 123 Authentication: 123 Password: Set SIP Registration: No SnomOne Trunk Type: SIP Gateway Account: 123 Domain: 192.168.10.250 Username: 123 Password: Set Proxy Address: 192.168.0.250 Send call to extension: 70 (tried both to the automated menu and a direct extension) Any advice on other settings to check? Thanks