I take it you didn't have to rem out callerid stuff and it gave you no errors? On May 5, 2012 10:08 PM, "Bryan Anderson" <[email protected]> wrote:
> 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 < > [email protected]> 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 <[email protected]> >> wrote: >> > ---------- Forwarded message ---------- >> > From: "Bryan Anderson" <[email protected]> >> > Date: May 5, 2012 3:37 PM >> > Subject: Re: [sipx-users] Patton Config from wiki >> > To: "Discussion list for users of sipXecs software" >> > <[email protected]> >> > >> > 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 < >> [email protected]> >> > wrote: >> >> >> >> You should put the latest version 6.1 on it. >> >> >> >> On May 5, 2012 1:31 AM, "Bryan Anderson" <[email protected]> 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 >> >>> <[email protected]> wrote: >> >>>> >> >>>> What firmware version? >> >>>> >> >>>> On May 4, 2012 5:50 PM, "Bryan Anderson" <[email protected]> >> 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 >> >>>>> [email protected] >> >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >>>> >> >>>> >> >>>> LAN/Telephony/Security and Control Systems Helpdesk: >> >>>> Telephone: 434.984.8426 >> >>>> sip: [email protected] >> >>>> >> >>>> Helpdesk Customers: http://myhelp.myitdepartment.net >> >>>> Blog: http://blog.myitdepartment.net >> >>>> >> >>>> _______________________________________________ >> >>>> sipx-users mailing list >> >>>> [email protected] >> >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >>> >> >>> >> >>> >> >>> _______________________________________________ >> >>> sipx-users mailing list >> >>> [email protected] >> >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> >> Telephone: 434.984.8426 >> >> sip: [email protected] >> >> >> >> Helpdesk Customers: http://myhelp.myitdepartment.net >> >> Blog: http://blog.myitdepartment.net >> >> >> >> _______________________________________________ >> >> sipx-users mailing list >> >> [email protected] >> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > >> > >> > >> > _______________________________________________ >> > sipx-users mailing list >> > [email protected] >> > List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> >> -- >> ~~~~~~~~~~~~~~~~~~ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] >> 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: [email protected] >> >> Helpdesk Customers: http://myhelp.myitdepartment.net >> Blog: http://blog.myitdepartment.net >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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