Thank you! thank you! thank you! thank you! It is up and working. I have set it to route properly where it is going. Now I can go drive 2 hours install it and drive 2 more hours to go home :)... Thanks Tony for your help with this and the grandstream. I am going to keep the grandstream and work with them tell they get it working or block all my emails and phone numbers :).
I do have a second of these Pattons here to learn with so that maybe next time it won't be quite so elementary. Thanks a lot, -Bryan Anderson On Sat, May 5, 2012 at 6:17 PM, Tony Graziano <tgrazi...@myitdepartment.net>wrote: > Try to adapt this one instead > > #----------------------------------------------------------------# > # # > # SN4524/JO/EUI # > # R6.1 2010-07-15 H323 SIP FXS FXO # > # 1970-07-02T18:35:24 # > # SN/00A0BA0505AA # > # Generated configuration file # > # # > #----------------------------------------------------------------# > > > cli version 3.20 > clock local default-offset -04:00 > dns-client server 192.168.54.2 > webserver port 80 language en > sntp-client server 192.5.41.40 > system hostname sip-gw.voice.mydomain.loc > > system > > ic voice 0 > low-bitrate-codec g729 > > profile ppp default > > profile call-progress-tone US_Dialtone > play 1 1000 350 -13 440 -13 > > profile call-progress-tone US_Alertingtone > play 1 2000 440 -19 480 -19 > pause 2 4000 > > profile call-progress-tone US_Busytone > play 1 500 480 -24 620 -24 > pause 2 500 > > profile tone-set default > profile tone-set US > map call-progress-tone dial-tone US_Dialtone > map call-progress-tone ringback-tone US_Alertingtone > map call-progress-tone busy-tone US_Busytone > map call-progress-tone release-tone US_Busytone > map call-progress-tone congestion-tone US_Busytone > > profile voip default > codec 1 g711alaw64k rx-length 20 tx-length 20 > codec 2 g711ulaw64k rx-length 20 tx-length 20 > > profile pstn default > > profile sip default > no autonomous-transitioning > > profile aaa default > method 1 local > method 2 none > > context ip router > > interface LAN > ipaddress 192.168.54.3 255.255.255.0 > tcp adjust-mss rx mtu > tcp adjust-mss tx mtu > > context ip router > route 0.0.0.0 0.0.0.0 192.168.54.1 > > context cs switch > digit-collection timeout 3 > > routing-table called-e164 SIP_TO_ISDN > route default dest-service OUTBOUND > > interface sip IF_SIPX > bind context sip-gateway GW-SIP > route call dest-table SIP_TO_ISDN > remote pbx.voice.mydomain.loc > address-translation outgoing-call to-header user-part fix 100 > host-part fix pbx.voice.mydomain.loc > > interface fxo IF_FXO0 > route call dest-interface IF_SIPX > disconnect-signal loop-break > disconnect-signal busy-tone > ring-number on-caller-id > dial-after timeout 2 > mute-dialing > use profile tone-set US > > interface fxo IF_FXO1 > route call dest-interface IF_SIPX > disconnect-signal loop-break > disconnect-signal busy-tone > ring-number on-caller-id > dial-after timeout 2 > mute-dialing > use profile tone-set US > > interface fxo IF_FXO2 > route call dest-interface IF_SIPX > disconnect-signal loop-break > disconnect-signal busy-tone > ring-number on-caller-id > dial-after timeout 2 > mute-dialing > use profile tone-set US > > interface fxo IF_FXO3 > route call dest-interface IF_SIPX > disconnect-signal loop-break > disconnect-signal busy-tone > ring-number on-caller-id > dial-after timeout 2 > mute-dialing > use profile tone-set US > > service hunt-group OUTBOUND > 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 > drop-cause user-busy > #route call 1 dest-interface IF_FXO3 > #route call 2 dest-interface IF_FXO2 > #route call 3 dest-interface IF_FXO1 > route call 3 dest-interface IF_FXO0 > > context cs switch > no shutdown > > location-service SIPX_SERVER > domain 1 sipx.voice.mydomain.loc > > context sip-gateway GW-SIP > > interface IF_SIPX > bind interface LAN context router port 5060 > > context sip-gateway GW-SIP > bind location-service SIPX_SERVER > no shutdown > > port ethernet 0 0 > medium auto > encapsulation ip > bind interface LAN router > no shutdown > > port ethernet 0 1 > medium 10 half > shutdown > > port fxo 0 0 > flash-hook-duration 50 > use profile fxo us > caller-id format bell > encapsulation cc-fxo > bind interface IF_FXO0 switch > no shutdown > > port fxo 0 1 > flash-hook-duration 50 > use profile fxo us > caller-id format bell > encapsulation cc-fxo > bind interface IF_FXO1 switch > shutdown > > port fxo 0 2 > flash-hook-duration 50 > use profile fxo us > caller-id format bell > encapsulation cc-fxo > bind interface IF_FXO2 switch > shutdown > > port fxo 0 3 > flash-hook-duration 50 > use profile fxo us > caller-id format bell > encapsulation cc-fxo > bind interface IF_FXO3 switch > shutdown > > When you upload the config, do a reload then do not save, it should > restart with the config you uploaded this way. If you save the config, > the one you have NOW overwrites the one you upload. (i.e., when it > asks you to drop changes, say yes). > > On Sat, May 5, 2012 at 8:30 PM, Bryan Anderson <branderso...@msn.com> > wrote: > > ---------- Forwarded message ---------- > > From: "Bryan Anderson" <shadow...@gmail.com> > > Date: May 5, 2012 3:37 PM > > Subject: Re: [sipx-users] Patton Config from wiki > > To: "Discussion list for users of sipXecs software" > > <sipx-users@list.sipfoundry.org> > > > > Ok, so I have the latest 6.1 on. I have attached my configuration. Out > > going is working but not incoming. The gateway is not answering the > calls. > > > > > > -Bryan Anderson > > > > > > > > > > On Sat, May 5, 2012 at 4:23 AM, Tony Graziano < > tgrazi...@myitdepartment.net> > > wrote: > >> > >> You should put the latest version 6.1 on it. > >> > >> On May 5, 2012 1:31 AM, "Bryan Anderson" <branderso...@msn.com> wrote: > >>> > >>> R5.2 > >>> > >>> I had gotten it to where some times it would call out. Now I dial the > >>> number, get a dial tone, then silence, then dial tone, then busy tone. > >>> > >>> -Bryan Anderson > >>> > >>> > >>> > >>> > >>> On Fri, May 4, 2012 at 7:05 PM, Tony Graziano > >>> <tgrazi...@myitdepartment.net> wrote: > >>>> > >>>> What firmware version? > >>>> > >>>> On May 4, 2012 5:50 PM, "Bryan Anderson" <branderso...@msn.com> > wrote: > >>>>> > >>>>> I have just received our two new Patton SN5420 4 FXO gateways. I > >>>>> pulled down the tested config from here: > >>>>> http://wiki.sipfoundry.org/display/sipXecs/Patton+4524+-+TESTED > >>>>> > >>>>> set the variables mentioned at the top of the page and removed the > two > >>>>> FXS port listing at the bottom of the config. Now I am getting boot > errors > >>>>> on reading the following four lines of the config. Please advise. > >>>>> > >>>>> context cs switch > >>>>> > >>>>> profile ringing-cadence default > >>>>> play 1 1000 > >>>>> pause 2 4000 > >>>>> > >>>>> > >>>>> > >>>>> -Bryan Anderson > >>>>> > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> sipx-users mailing list > >>>>> sipx-users@list.sipfoundry.org > >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >>>> > >>>> > >>>> LAN/Telephony/Security and Control Systems Helpdesk: > >>>> Telephone: 434.984.8426 > >>>> sip: helpd...@voice.myitdepartment.net > >>>> > >>>> Helpdesk Customers: http://myhelp.myitdepartment.net > >>>> Blog: http://blog.myitdepartment.net > >>>> > >>>> _______________________________________________ > >>>> sipx-users mailing list > >>>> sipx-users@list.sipfoundry.org > >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >>> > >>> > >>> > >>> _______________________________________________ > >>> sipx-users mailing list > >>> sipx-users@list.sipfoundry.org > >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >> > >> > >> LAN/Telephony/Security and Control Systems Helpdesk: > >> Telephone: 434.984.8426 > >> sip: helpd...@voice.myitdepartment.net > >> > >> Helpdesk Customers: http://myhelp.myitdepartment.net > >> Blog: http://blog.myitdepartment.net > >> > >> _______________________________________________ > >> sipx-users mailing list > >> sipx-users@list.sipfoundry.org > >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > > > > _______________________________________________ > > sipx-users mailing list > > sipx-users@list.sipfoundry.org > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgrazi...@voice.myitdepartment.net > Fax: 434.465.6833 > ~~~~~~~~~~~~~~~~~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~~~~~~~~~~~~~~~~~ > > -- > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpd...@voice.myitdepartment.net > > Helpdesk Customers: http://myhelp.myitdepartment.net > Blog: http://blog.myitdepartment.net > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >
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