I meant to attach the configuration on the last email, here it is:

fxo2.foo21.com#show running-config
#----------------------------------------------------------------#
#                                                                #
# SN4114/JO/EUI                                                  #
# R5.4 2009-11-18 H323 SIP FXS FXO                               #
# 2010-01-24T21:18:14                                            #
# SN/00A0BA04EB81                                                #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.20
clock local offset -06:00
dns-client server 192.172.252.1
webserver port 80 language en
sntp-client
sntp-client server primary 192.172.252.1 port 123 version 4
sntp-client poll-interval 36000
system hostname fxo2.foo21.com

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_Altertingtone
 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 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 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default

profile aaa default
 method 1 local
 method 2 none

context ip router

 interface eth0
   ipaddress 192.172.252.22 255.255.255.0
   tcp adjust-mss rx mtu
   tcp adjust-mss tx mtu

context ip router
 route 0.0.0.0 0.0.0.0 192.172.252.1 0

context cs switch
 digit-collection timeout 4

 routing-table called-e164 SIP-TO-PP
   route default dest-service PP-HUNT

 mapping-table calling-e164 to calling-uri CID_MAP
   map default to [email protected]

 interface sip IF-SIP-PP
   bind context sip-gateway SIP-GW-PP
   route call dest-table SIP-TO-PP
   remote 192.172.252.20 5060
   address-translation outgoing-call to-header user-part fix 0 host-part fix 
192.172.252.20
   use profile tone-set US

 interface fxo IF-FXO0
   route call dest-interface IF-SIP-PP
   disconnect-signal loop-break
   disconnect-signal busy-tone
   ring-number on-caller-id
   mute-dialing
   use profile tone-set US

 service hunt-group PP-HUNT
   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-FXO0

context cs switch
 no shutdown

location-service SIPX-SERVER
 domain 1 192.172.252.20 5060

context sip-gateway SIP-GW-PP

 interface IF-IP
   bind interface eth0 context router port 5060

context sip-gateway SIP-GW-PP
 no shutdown

port ethernet 0 0
 medium auto
 encapsulation ip
 bind interface eth0 router
 no 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
 shutdown

port fxo 0 2
 shutdown

port fxo 0 3
 shutdown



On Jan 24, 2010, at 9:21 PM, Eric Varsanyi wrote:

> I have a Patton 4114 FXO box and its working well but I'd like to change the 
> 'From' URI from 'anonymous' if no CID is received to something else (the 
> depends on the FXO interface the call was received on). I tried creating a 
> mapping table in the cs switch context:
> 
> mapping-table calling-e164 to calling-uri CID_MAP
>   map default to [email protected]
> 
> However I still get From: [email protected] (where the nn's are the 
> address of the patton).
> 
> I suspect I'm just not getting how to attach this mapping table to the 
> (implied?) routing table that handles called from the FXO to the SIP gateway 
> in the call router.
> 
> Any hints would be appreciated, I have 3 different classes of 'anonymous' I'd 
> like to categorize based on which FXO port the call comes in on.
> 
> Thanks,
> -Eric Varsanyi
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