shopcomputer

The audiocodes will be MP-11X, you can follow the Wiki to configure it.

This should work for Broadvox, unless they are overwriting the caller ID on their end. Global: yes Trunk ANI: 5555551212 Remote Party/Privacy Indication: Remote-Party-ID

What do you have set for the ANI on the trunk?

Set the trunks go forward to a hunt group, add the receptionist to the hunt group. Set the service flag on the hunt group.

Scroll down, you will see the 3.2 release updates.

Yes we sell the cyberdata it works great, gets set up as an extension on the PBX.

You can easily confirm it is on a public IP, just by logging in remotely and doing an ipconfig from a command prompt.

Which Firewall do you have at the server location? Is the server behind NAT?

Which Firewall do you have at the server location? Is the server behind NAT?

This does work Vitelity, junction networks and Broadvox.

Click system, PSTN-Gateway, Detect Polarity Change, set it to off.

As I stated above caller ID is working correctly, my problem is that the caller ID issue http://wiki.pbxnsip.com/index.php/Release_..._3.1#Cell_Phone does not seem to be working for me, it still displays the cell phone number as the caller ID.

When placing an outbound call using a cell phone, it is still showing the cell phone number as the caller ID, I thought this was supposed to be fixed in this version. 3.1.1.3113 (Win32), I tied with 2 carriers, Broadvox and Vitelity, all other caller ID functions are working correctly, does it have to do with Remote Party/Privacy Indication:? I tried several setting they all show the cell phone number as the caller ID, except no indication, which shows unavailabe. [7] 2008/12/25 11:23:17: SIP Rx udp:64.2.142.30:5060: INVITE sip:2124007000@192.168.10.2 SIP/2.0 Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c To: <sip:2124007000@216.112.126.83> Contact: <sip:9175551212@64.2.142.30> Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Dec 2008 16:23:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 306 v=0 o=root 4115 4115 IN IP4 64.2.142.30 s=session c=IN IP4 64.2.142.30 t=0 0 m=audio 11768 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - [9] 2008/12/25 11:23:17: UDP: Opening socket on port 58244 [9] 2008/12/25 11:23:17: UDP: Opening socket on port 58245 [5] 2008/12/25 11:23:17: Identify trunk (IP address and DID match) 5 [9] 2008/12/25 11:23:17: Resolve 50145: aaaa udp 64.2.142.30 5060 [9] 2008/12/25 11:23:17: Resolve 50145: a udp 64.2.142.30 5060 [9] 2008/12/25 11:23:17: Resolve 50145: udp 64.2.142.30 5060 [7] 2008/12/25 11:23:17: SIP Tx udp:64.2.142.30:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport=5060 From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c To: <sip:2124007000@216.112.126.83>;tag=09c02f6724 Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30 CSeq: 102 INVITE Content-Length: 0 [6] 2008/12/25 11:23:17: Sending RTP for 29f00291573171d169e7ada9613cbd6a@64.2.142.30#09c02f6724 to 64.2.142.30:11768 [5] 2008/12/25 11:23:17: Trunk vitelity inbound sends call to +12124007000 in domain localhost [7] 2008/12/25 11:23:17: Received call from cell phone +19175551212 [8] 2008/12/25 11:23:17: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50 [9] 2008/12/25 11:23:17: Resolve 50146: aaaa udp 64.2.142.30 5060 [9] 2008/12/25 11:23:17: Resolve 50146: a udp 64.2.142.30 5060 [9] 2008/12/25 11:23:17: Resolve 50146: udp 64.2.142.30 5060 [7] 2008/12/25 11:23:17: SIP Tx udp:64.2.142.30:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport=5060 From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c To: <sip:2124007000@216.112.126.83>;tag=09c02f6724 Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30 CSeq: 102 INVITE Contact: <sip:torn_pbx@192.168.10.2:5060;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/3.1.1.3113 Content-Type: application/sdp Content-Length: 286 v=0 o=- 22539 22539 IN IP4 192.168.10.2 s=- c=IN IP4 192.168.10.2 t=0 0 m=audio 58244 RTP/AVP 0 8 18 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2008/12/25 11:23:17: Resolve 50147: aaaa udp 64.2.142.30 5060 [9] 2008/12/25 11:23:17: Resolve 50147: a udp 64.2.142.30 5060 [9] 2008/12/25 11:23:17: Resolve 50147: udp 64.2.142.30 5060 [7] 2008/12/25 11:23:17: SIP Tx udp:64.2.142.30:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK02616f7b;rport=5060 From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c To: <sip:2124007000@216.112.126.83>;tag=09c02f6724 Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30 CSeq: 102 INVITE Contact: <sip:torn_pbx@192.168.10.2:5060;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/3.1.1.3113 Content-Type: application/sdp Content-Length: 286 v=0 o=- 22539 22539 IN IP4 192.168.10.2 s=- c=IN IP4 192.168.10.2 t=0 0 m=audio 58244 RTP/AVP 0 8 18 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2008/12/25 11:23:17: SIP Rx udp:64.2.142.30:5060: ACK sip:torn_pbx@192.168.10.2:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK249a5232;rport From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c To: <sip:2124007000@216.112.126.83>;tag=09c02f6724 Contact: <sip:9175551212@64.2.142.30> Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [7] 2008/12/25 11:23:17: SIP Rx udp:64.2.142.30:5060: ACK sip:torn_pbx@192.168.10.2:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 64.2.142.30:5060;branch=z9hG4bK306ceaaf;rport From: "9175551212" <sip:9175551212@64.2.142.30>;tag=as65516e2c To: <sip:2124007000@216.112.126.83>;tag=09c02f6724 Contact: <sip:9175551212@64.2.142.30> Call-ID: 29f00291573171d169e7ada9613cbd6a@64.2.142.30 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [9] 2008/12/25 11:23:17: Message repetition, packet dropped [6] 2008/12/25 11:23:20: Received DTMF 1 [8] 2008/12/25 11:23:20: Play audio_en/ex_enter_access_code.wav [8] 2008/12/25 11:23:22: Play space20 [6] 2008/12/25 11:23:23: Received DTMF 1 [6] 2008/12/25 11:23:23: Received DTMF 9 [7] 2008/12/25 11:23:24: SIP Rx udp:66.114.64.65:2048: SUBSCRIBE sip:401@sip.solutions.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-7vbhqhxh0w98;rport From: <sip:401@sip.solutions.com>;tag=gren4n4yrj To: <sip:401@sip.solutions.com;user=phone>;tag=aad9a04baf Call-ID: 3c26701c35a2-j1weli7digqg CSeq: 2533 SUBSCRIBE Max-Forwards: 70 Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;reg-id=1 Event: message-summary Accept: application/simple-message-summary User-Agent: snom360/7.3.10a Proxy-Require: buttons Expires: 3600 Content-Length: 0 [9] 2008/12/25 11:23:24: Resolve 50148: udp 66.114.64.65 2048 [7] 2008/12/25 11:23:24: SIP Tx udp:66.114.64.65:2048: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-7vbhqhxh0w98;rport=2048;received=66.114.64.65 From: <sip:401@sip.solutions.com>;tag=gren4n4yrj To: <sip:401@sip.solutions.com;user=phone>;tag=aad9a04baf Call-ID: 3c26701c35a2-j1weli7digqg CSeq: 2533 SUBSCRIBE Contact: <sip:192.168.10.2:5060;transport=udp> Expires: 32 Content-Length: 0 [7] 2008/12/25 11:23:24: SIP Rx udp:66.114.64.65:2048: REGISTER sip:sip.solutions.com SIP/2.0 Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-txke68p6cw17;rport From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=fkb206hmbx To: "Moishe Grunstein" <sip:401@sip.solutions.com> Call-ID: 3c26701b6b59-zzy3nchymw3e CSeq: 5022 REGISTER Max-Forwards: 70 Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:7035dcbf-7d47-44e4-9f8a-06b3fb57c1b6>" Contact: <http://192.168.81.102:80> Contact: <https://192.168.81.102:443> User-Agent: snom360/7.3.10a Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.81.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [9] 2008/12/25 11:23:24: Resolve 50149: udp 66.114.64.65 2048 [7] 2008/12/25 11:23:24: SIP Tx udp:66.114.64.65:2048: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-txke68p6cw17;rport=2048;received=66.114.64.65 From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=fkb206hmbx To: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=673192d660 Call-ID: 3c26701b6b59-zzy3nchymw3e CSeq: 5022 REGISTER Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;expires=28 Contact: <http://192.168.81.102:80>;expires=28 Contact: <https://192.168.81.102:443>;expires=28 Content-Length: 0 [6] 2008/12/25 11:23:24: Received DTMF 7 [6] 2008/12/25 11:23:25: Received DTMF 6 [8] 2008/12/25 11:23:25: Play audio_en/ex_enter_number.wav [6] 2008/12/25 11:23:27: Received DTMF 7 [6] 2008/12/25 11:23:28: Received DTMF 1 [6] 2008/12/25 11:23:29: Received DTMF 8 [6] 2008/12/25 11:23:29: Received DTMF 4 [6] 2008/12/25 11:23:30: Received DTMF 3 [6] 2008/12/25 11:23:31: Received DTMF 6 [6] 2008/12/25 11:23:32: Received DTMF 5 [6] 2008/12/25 11:23:32: Received DTMF 5 [6] 2008/12/25 11:23:34: Received DTMF 5 [6] 2008/12/25 11:23:35: Received DTMF 5 [6] 2008/12/25 11:23:38: Received DTMF # [9] 2008/12/25 11:23:38: Dialplan: Evaluating !^(9411)@.*!sip:18005558355@\r;user=phone!i against 7184365555@localhost [9] 2008/12/25 11:23:38: Dialplan: Evaluating !^([0-9]{7})@.*!sip:\1@\r;user=phone!i against 7184365555@localhost [9] 2008/12/25 11:23:38: Dialplan: Evaluating !^([0-9]{10})@.*!sip:1\1@\r;user=phone!i against 7184365555@localhost [5] 2008/12/25 11:23:38: Dialplan outbound: Match 7184365555@localhost to <sip:17184365555@outbound1.vitelity.net;user=phone> on trunk vitelity outbound [8] 2008/12/25 11:23:38: Play audio_moh/noise.wav [9] 2008/12/25 11:23:38: UDP: Opening socket on port 62514 [9] 2008/12/25 11:23:38: UDP: Opening socket on port 62515 [9] 2008/12/25 11:23:38: Resolve 50150: url sip:outbound1.vitelity.net [9] 2008/12/25 11:23:38: Resolve 50150: naptr outbound1.vitelity.net [8] 2008/12/25 11:23:38: DNS: Add dns_naptr outbound1.vitelity.net (ttl=60) [9] 2008/12/25 11:23:38: Resolve 50150: naptr outbound1.vitelity.net [9] 2008/12/25 11:23:38: Resolve 50150: srv tls _sips._tcp.outbound1.vitelity.net [8] 2008/12/25 11:23:38: DNS: Add dns_srv _sips._tcp.outbound1.vitelity.net (ttl=60) [9] 2008/12/25 11:23:38: Resolve 50150: srv tls _sips._tcp.outbound1.vitelity.net [9] 2008/12/25 11:23:38: Resolve 50150: srv tcp _sip._tcp.outbound1.vitelity.net [8] 2008/12/25 11:23:38: DNS: Add dns_srv _sip._tcp.outbound1.vitelity.net (ttl=60) [9] 2008/12/25 11:23:38: Resolve 50150: srv tcp _sip._tcp.outbound1.vitelity.net [9] 2008/12/25 11:23:38: Resolve 50150: srv udp _sip._udp.outbound1.vitelity.net [7] 2008/12/25 11:23:38: SIP Rx udp:66.114.64.65:2048: REGISTER sip:sip.solutions.com SIP/2.0 Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-g6b5c7lxejtn;rport From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=hy6e1yqunm To: "Moishe Grunstein" <sip:401@sip.solutions.com> Call-ID: 3c26701b6b59-zzy3nchymw3e CSeq: 5023 REGISTER Max-Forwards: 70 Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:7035dcbf-7d47-44e4-9f8a-06b3fb57c1b6>" Contact: <http://192.168.81.102:80> Contact: <https://192.168.81.102:443> User-Agent: snom360/7.3.10a Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.81.102 Proxy-Require: buttons Expires: 3600 Content-Length: 0 [9] 2008/12/25 11:23:38: Resolve 50151: udp 66.114.64.65 2048 [7] 2008/12/25 11:23:38: SIP Tx udp:66.114.64.65:2048: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.81.102:2048;branch=z9hG4bK-g6b5c7lxejtn;rport=2048;received=66.114.64.65 From: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=hy6e1yqunm To: "Moishe Grunstein" <sip:401@sip.solutions.com>;tag=673192d660 Call-ID: 3c26701b6b59-zzy3nchymw3e CSeq: 5023 REGISTER Contact: <sip:401@192.168.81.102:2048;line=7uuu69wp>;expires=32 Contact: <http://192.168.81.102:80>;expires=32 Contact: <https://192.168.81.102:443>;expires=32 Content-Length: 0 [8] 2008/12/25 11:23:38: DNS: Add dns_srv _sip._udp.outbound1.vitelity.net (ttl=60) [9] 2008/12/25 11:23:38: Resolve 50150: srv udp _sip._udp.outbound1.vitelity.net [9] 2008/12/25 11:23:38: Resolve 50150: a udp outbound1.vitelity.net 5060 [8] 2008/12/25 11:23:38: DNS: Add dns_a outbound1.vitelity.net 64.2.142.87 (ttl=864) [9] 2008/12/25 11:23:38: Resolve 50150: a udp outbound1.vitelity.net 5060 [9] 2008/12/25 11:23:38: Resolve 50150: udp 64.2.142.86 5060 [7] 2008/12/25 11:23:38: SIP Tx udp:64.2.142.86:5060: INVITE sip:17184365555@outbound1.vitelity.net;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK-cc1c0f10a8de9fd8f30569e582ffce5f;rport From: "9175551212" <sip:9175551212@localhost;user=phone>;tag=58440 To: <sip:17184365555@outbound1.vitelity.net;user=phone> Call-ID: 60d6e30b@pbx CSeq: 23051 INVITE Max-Forwards: 70 Contact: <sip:torn_pbx@192.168.10.2:5060;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/3.1.1.3113 P-Asserted-Identity: <sip:torn_pbx@outbound1.vitelity.net> Content-Type: application/sdp Content-Length: 335 v=0 o=- 9354 9354 IN IP4 192.168.10.2 s=- c=IN IP4 192.168.10.2 t=0 0 m=audio 62514 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no 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] 2008/12/25 11:23:38: SIP Rx udp:64.2.142.20:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK-cc1c0f10a8de9fd8f30569e582ffce5f;received=216.112.126.83;rport=5060 From: "9175551212" <sip:9175551212@localhost;user=phone>;tag=58440 To: <sip:17184365555@outbound1.vitelity.net;user=phone> Call-ID: 60d6e30b@pbx CSeq: 23051 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:17184365555@64.2.142.20> Content-Length: 0 [7] 2008/12/25 11:23:39: SIP Rx udp:74.64.70.251:1323: REGISTER sip:sip.solutions.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253:5060;rport;branch=z9hG4bK2126580982 From: <sip:420@sip.solutions.com>;tag=518775785 To: <sip:420@sip.solutions.com> Call-ID: 801557023@192.168.0.253 CSeq: 919 REGISTER Contact: <sip:420@192.168.0.253:5060> Max-Forwards: 5 User-Agent: Linphone-1.1.0 MX-Video/eXosip Expires: 200 Content-Length: 0 [9] 2008/12/25 11:23:39: Resolve 50152: udp 74.64.70.251 1323 [7] 2008/12/25 11:23:39: SIP Tx udp:74.64.70.251:1323: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 192.168.0.253:5060;rport=1323;branch=z9hG4bK2126580982;received=74.64.70.251 From: <sip:420@sip.solutions.com>;tag=518775785 To: <sip:420@sip.solutions.com>;tag=b882f81be3 Call-ID: 801557023@192.168.0.253 CSeq: 919 REGISTER User-Agent: pbxnsip-PBX/3.1.1.3113 WWW-Authenticate: Digest realm="sip.solutions.com",nonce="8313bfe2108fc54eadd0c0ec083443e5",domain="sip:sip.solutions.com",algorithm=MD5 Content-Length: 0 [7] 2008/12/25 11:23:40: SIP Rx udp:64.2.142.20:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK-cc1c0f10a8de9fd8f30569e582ffce5f;received=216.112.126.83;rport=5060 From: "9175551212" <sip:9175551212@localhost;user=phone>;tag=58440 To: <sip:17184365555@outbound1.vitelity.net;user=phone>;tag=as0bb732e3 Call-ID: 60d6e30b@pbx CSeq: 23051 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:17184365555@64.2.142.20> Content-Length: 0 [8] 2008/12/25 11:23:40: Play audio_en/ringback.wav

Firewall?

I have a system with call center edition license. The Snom ad hoc recording, is sending the recording to the mailbox, however it is not emailing it me, the way it does with all the voicemails. This used to work a while back. I am not sure when it stopped as I rarely use it. I do see this in the PBX log. User-Agent: snom360/7.3.10a Record: on Proxy-Require: buttons Content-Length: 0 and User-Agent: pbxnsip-PBX/3.1.1.3113 Record: on Content-Length: 0

I think it is a bug, I just had a customer email me, it calls her outside the specified hours too.

.

Why don't we create a browse button to upload files, the way you have it for autoattendants and IVR nodes?

Yes I set my remote phone back to manual configuration, routing rules, were causing RTP no audio issues, I just was playing with it, this is in my demo system that is behind a CiSCO Pix in a data center. Their SIP ALG works good, when the phone is manually configured.

Whic phones are you using, in our Snom phones, it show the agent group name the caller reached, plus his original caller ID.

I had localhost, changed to IP, did not change the PnP, although routing list, under ports, seems to change the PnP IP.

I think you are referring to the domain name, that has seems to have no effect on what IP the provisiong provides. All users are remote in this case.

What do you mean by domain IP?

I am trying to provision a snom phone, where the pbx is behind NAT. The SIP IP replacement does not work for the htttp provisioning it seems. Is there anywhere I can set an ip replacement for the provisioning? http://server_public_address/provisioning/...-macaddress.htm returns the private ip <?xml version="1.0" encoding="utf-8" ?> - <setting-files> <file url="http://192.168.10.2:8000/provisioning/snom_3xx_phone-000413290587.xml?model=snom360" /> <file url="http://192.168.10.2:8000/provisioning/snom_3xx_fkeys-000413290587.xml" /> <file url="http://192.168.10.2:8000/provisioning/snom_web_lang.xml" /> <file url="http://192.168.10.2:8000/provisioning/snom_gui_lang.xml" /> </setting-files>

The need for the Pin code is very important, otherwise. I can set my system to show your number as the caller ID, then I can call in and use your system, for long distance calls. I can also check any sprint cell phones voicemail unless they set it to use a pin.