I did have to move the caller-id from the ports to interfaces other wise I got keyword mismatch, but all worked without issue. It's in place and I didn't see anything in email regarding issues, but I am on vacation finally after three emergency sipxecs installations in a row.
Thanks again. On May 6, 2012 4:15 AM, "Tony Graziano" <tgrazi...@myitdepartment.net> wrote: > > 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" <branderso...@msn.com> 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 < 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/ >> >> >> >> _______________________________________________ >> 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/
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