halalabu Posted May 2, 2012 Report Share Posted May 2, 2012 Hi Folks, I'm trying to set up a Patton 4114 as an FXO gateway connected to POTS lines. I have my Snom set up and working internally, but when I try to dial through the gateway, i get the error "not in service, please check the number and dial again". The off-hook light does light up on the 4114 when I place a call, so it is communicating at some level. I feel like I'm missing something simple but I don't know what it is. Any help would be greatly appreciated. Just an FYI you'll see the 4114 is configured for DHCP, I have a reservation set on the router for its MAC address, so it is a static address. Thanks so much! Hal SNOM PBX, SNTP Server, DNS Server, are all located at 10.0.0.202 (also services.alabu.com) Router/Gateway are located at 10.0.0.1 Patton 4114 is located at 10.0.0.203 Network Equipment: Cisco SRP541W Router Cisco SG300 Managed Switch Mac Mini Server running OS X 10.7 Lion Here is my Patton configuration: #################################################################################### # Configuration file for SmartNode with: 1 FXO and 0 FXS # With one ethernet port # Configuration file name is alabu-fxo # Snome One IPPBX #################################################################################### # LAN Interface Data # IP Address dhcp # IP Mask #################################################################################### # WAN Interface Data # IP Address dhcp # IP Mask #################################################################################### # Default Gateway 10.0.0.1 #################################################################################### # IP-PBX/SIP PROXY DATA # IP-PBX Ip Address / Name / Domain services.alabu.com # IP-PBX SIP Port 5060 #################################################################################### # LOCALIZATION DATA # Country United States #################################################################################### # Supported firmware versions R5.x # Tool version V2_1 06222011 # Adapted: Ernesto Casas # for comments, feedback or suggestions please send an email to: ecasas@patton.com #################################################################################### #################################################################################### webserver port 80 language en sntp-client server 10.0.0.202 dns-client server 10.0.0.202 system ic voice 0 low-bitrate-codec g729 profile ppp default # Call Progress tones based on country selection. Your selected country is: United States profile call-progress-tone defaultDialtone flush-play-list play 1 1000 350 -13 440 -13 profile call-progress-tone defaultAlertingtone flush-play-list play 1 1000 440 -19 480 -19 pause 2 3000 profile call-progress-tone defaultBusytone flush-play-list play 1 500 480 -24 620 -24 pause 2 500 profile call-progress-tone defaultReleasetone flush-play-list play 1 250 480 -24 620 -24 pause 2 250 profile call-progress-tone defaultCongestiontone flush-play-list play 1 250 480 -24 620 -24 pause 2 250 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 10.0.0.1 context cs switch digit-collection timeout 2 interface sip IF_SIP_1 bind context sip-gateway GW_SIP_ALL_LINES no early-connect early-disconnect route call dest-service HUNT_FXO remote services.alabu.com 5060 address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none address-translation incoming-call called-e164 request-uri interface fxo IF_FXO_1 route call dest-interface IF_SIP_1 loop-break-duration min 200 max 1000 ring-number on-caller-id mute-dialing disconnect-signal loop-break disconnect-signal busy-tone dial-after timeout 1 service hunt-group HUNT_FXO cyclic 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 context cs switch no shutdown authentication-service AS_ALL_LINES username 10002 password 10002 location-services LS_ALL_LINES identity 10002 port fxo 0 0 use profile fxo us encapsulation cc-fxo bind interface IF_FXO_1 switch no shutdown 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 ################################################# END ################################################## Here is my Snom Configuration. Here is my Snom logs after trying to place one call. I replaced the phone number dialed with "X"s. [5] 2012/05/02 12:46:18: SIP Rx tls:10.0.0.101:4864: INVITE sip:XXXXXXX@services.alabu.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone> Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1 X-Serialnumber: 0004134535DF P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.18 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 518 v=0 o=root 893378661 893378661 IN IP4 10.0.0.101 s=call c=IN IP4 10.0.0.101 t=0 0 m=audio 60964 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7K0KKEDkZy8CGgPPJpamWwWhwLYVlW1Ovwh8/LyP a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [1] 2012/05/02 12:46:18: UDP: TOS could not be set [1] 2012/05/02 12:46:18: Last message repeated 2 times [5] 2012/05/02 12:46:18: SIP Rx tls:10.0.0.101:4864: INVITE sip:XXXXXXX@services.alabu.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone> Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1 X-Serialnumber: 0004134535DF P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.18 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 518 v=0 o=root 893378661 893378661 IN IP4 10.0.0.101 s=call c=IN IP4 10.0.0.101 t=0 0 m=audio 60964 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7K0KKEDkZy8CGgPPJpamWwWhwLYVlW1Ovwh8/LyP a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt a=sendrecv [5] 2012/05/02 12:46:18: SIP Tx tls:10.0.0.101:4864: SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport=4864 From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 1 INVITE Content-Length: 0 [5] 2012/05/02 12:46:18: Dialplan "Alabu Skin Care": Match 518XXXXXXX@services.alabu.com to <sip:518XXXXXXX@10.0.0.203:5060;user=phone> on trunk FXO 0 [1] 2012/05/02 12:46:18: UDP: TOS could not be set [1] 2012/05/02 12:46:18: Last message repeated 2 times [5] 2012/05/02 12:46:18: SIP Tx udp:10.0.0.203:5060: INVITE sip:518XXXXXXX@10.0.0.203:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-4c308048256b66b1a6c4606b9073cb7a;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298 To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone> Call-ID: 0a73e210@pbx CSeq: 10349 INVITE Max-Forwards: 70 Contact: <sip:75902@10.0.0.202:5060;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/2011-4.2.0.3981 P-Asserted-Identity: "FXO 0" <sip:75902@services.alabu.com> Content-Type: application/sdp Content-Length: 333 v=0 o=- 1035386651 1035386651 IN IP4 10.0.0.202 s=- c=IN IP4 10.0.0.202 t=0 0 m=audio 58570 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 [5] 2012/05/02 12:46:18: SIP Tx tls:10.0.0.101:4864: SIP/2.0 183 Session Progress Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport=4864 From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 1 INVITE Contact: <sip:75902@10.0.0.202:5081;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 429 v=0 o=- 1654920880 1654920880 IN IP4 10.0.0.202 s=- c=IN IP4 10.0.0.202 t=0 0 m=audio 58242 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aEgAj6nBDSI9lfuXjfuoFknhRrbgteBut8zJ8MW7 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=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/05/02 12:46:18: SIP Rx udp:10.0.0.203:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-4c308048256b66b1a6c4606b9073cb7a;rport=5060;received=10.0.0.202 From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298 To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone> Call-ID: 0a73e210@pbx CSeq: 10349 INVITE Server: Patton SN4114 JO EUI 00A0BA071F53 R6.1 2012-03-07 H323 SIP FXS FXO M5T SIP Stack/4.0.30.30 Content-Length: 0 [5] 2012/05/02 12:46:19: SIP Rx tls:10.0.0.101:4864: PRACK sip:75902@10.0.0.202:5081;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-92icuy4qau22;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [5] 2012/05/02 12:46:19: SIP Tx tls:10.0.0.101:4864: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-92icuy4qau22;rport=4864 From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 2 PRACK Contact: <sip:75902@10.0.0.202:5081;transport=tls> User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2012/05/02 12:46:22: SIP Rx udp:10.0.0.203:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-4c308048256b66b1a6c4606b9073cb7a;rport=5060;received=10.0.0.202 From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298 To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227 Call-ID: 0a73e210@pbx CSeq: 10349 INVITE Contact: <sip:518XXXXXXX@10.0.0.203:5060> Server: Patton SN4114 JO EUI 00A0BA071F53 R6.1 2012-03-07 H323 SIP FXS FXO M5T SIP Stack/4.0.30.30 Supported: replaces Content-Type: application/sdp Content-Length: 215 v=0 o=MxSIP 0 10 IN IP4 10.0.0.203 s=SIP Call c=IN IP4 10.0.0.203 t=0 0 m=audio 4872 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 [5] 2012/05/02 12:46:22: SIP Tx udp:10.0.0.203:5060: ACK sip:518XXXXXXX@10.0.0.203:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-a9c127f03d3880e3dab8fef72925ef42;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298 To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227 Call-ID: 0a73e210@pbx CSeq: 10349 ACK Max-Forwards: 70 Contact: <sip:75902@10.0.0.202:5060;transport=udp> P-Asserted-Identity: "FXO 0" <sip:75902@services.alabu.com> Content-Length: 0 [5] 2012/05/02 12:46:22: SIP Tx tls:10.0.0.101:4864: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport=4864 From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 1 INVITE Contact: <sip:75902@10.0.0.202:5081;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Type: application/sdp Content-Length: 429 v=0 o=- 1654920880 1654920880 IN IP4 10.0.0.202 s=- c=IN IP4 10.0.0.202 t=0 0 m=audio 58242 RTP/AVP 0 8 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aEgAj6nBDSI9lfuXjfuoFknhRrbgteBut8zJ8MW7 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=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/05/02 12:46:22: SIP Rx tls:10.0.0.101:4864: ACK sip:75902@10.0.0.202:5081;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-rfv1on8evf71;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1 Proxy-Require: buttons Content-Length: 0 [5] 2012/05/02 12:46:24: SIP Rx tls:10.0.0.101:4864: BYE sip:75902@10.0.0.202:5081;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-c742rpx0rfw5;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1 User-Agent: snom821/8.4.18 RTP-RxStat: Total_Rx_Pkts=272,Rx_Pkts=265,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=267,Tx_Pkts=267,Remote_Tx_Pkts=250 Proxy-Require: buttons Content-Length: 0 [5] 2012/05/02 12:46:24: SIP Tx tls:10.0.0.101:4864: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-c742rpx0rfw5;rport=4864 From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625 Call-ID: 189b273cef94-lwpfxubupkbu CSeq: 3 BYE Contact: <sip:75902@10.0.0.202:5081;transport=tls> User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2012/05/02 12:46:24: SIP Tx udp:10.0.0.203:5060: BYE sip:518XXXXXXX@10.0.0.203:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-50b51dc4e9ad2ad0b2059ae4e803fb20;rport From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298 To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227 Call-ID: 0a73e210@pbx CSeq: 10350 BYE Max-Forwards: 70 Contact: <sip:75902@10.0.0.202:5060;transport=udp> P-Asserted-Identity: "FXO 0" <sip:75902@services.alabu.com> Content-Length: 0 [5] 2012/05/02 12:46:24: SIP Rx udp:10.0.0.203:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-50b51dc4e9ad2ad0b2059ae4e803fb20;rport=5060;received=10.0.0.202 From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298 To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227 Call-ID: 0a73e210@pbx CSeq: 10350 BYE Server: Patton SN4114 JO EUI 00A0BA071F53 R6.1 2012-03-07 H323 SIP FXS FXO M5T SIP Stack/4.0.30.30 Content-Length: 0 [5] 2012/05/02 12:46:24: BYE Response: Terminate 0a73e210@pbx Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 3, 2012 Report Share Posted May 3, 2012 The SIP log looks good/beautiful. To me it seems that the carrier is generating the prompt. Maybe there is a "1" missing at the beginning of the number? Try to connect a analog phone to the line, dial the number through the analog line and see if you get the same answer. Quote Link to comment Share on other sites More sharing options...
Steve B Posted May 3, 2012 Report Share Posted May 3, 2012 I would get this error if the port didn't have dial tone. Check for dial tone on port 1 first...and I have some notes below: You have the gateway set to dhcp and the server set to services.alabu.com 5060. I would first give the adapter a static IP (if it doesn't have a dedicated DHCP address, something like 10.0.0.41 255.255.255.0), and I see you have the PBX pointing to an internal IP for the gateway so I assume the internal IP of the PBX is a 10.0.0.x IP? Point the gateway there. You seem to have set up an identity: interface sip IF_SIP_1 bind context sip-gateway GW_SIP_ALL_LINES no early-connect early-disconnect route call dest-service HUNT_FXO remote services.alabu.com 5060 address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none address-translation incoming-call called-e164 request-uri location-services LS_ALL_LINES identity 10002 I don't know what this does but my setup didn't have it. If the above steps didn't work, try deleting the 10002 from both areas. Everything else seems to be the same way I set my config file up. Also, set the Explicitly list address for inbound traffic in the PBX trunk after you get it up and running, if you don't it leaves the system open for someone else to use try and use your PBX. Quote Link to comment Share on other sites More sharing options...
halalabu Posted May 18, 2012 Author Report Share Posted May 18, 2012 I contacted Patton support and they got me squared away. There were multiple config issues with the Patton box, and also my local phone company is very strict with how the numbers are dialed. Thanks for your help! Quote Link to comment Share on other sites More sharing options...
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