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
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