Hi Michael!

The attached config works fine with me!

regards,
klaus

-------------------------------------------------------------
Running configuration:
#----------------------------------------------------------------#
#                                                                #
# SN1200                                                         #
# R3.10 BUILD21288 SIP                                           #
# 2001-02-11T04:13:39                                            #
# Generated configuration file                                   #
#                                                                #
#----------------------------------------------------------------#

cli version 3.00
administrator root password tGMDasdfvledtQbE1agA== encrypted
dns-client server 195.58.160.2
dns-client cache 30
snmp community public ro
sntp-client server primary 193.171.3.1 port 123 version 4
sntp-client server secondary 131.130.1.11 port 123 version 4
sntp-client anycast-address 224.0.1.1 port 123

profile ppp default

profile tone-set default

profile voip de

profile voip myvoip
  codec 1 g711alaw64k rx-length 20 tx-length 20
  codec 2 g711ulaw64k rx-length 20 tx-length 20
  codec 3 g729 rx-length 20 tx-length 20

profile pstn default

profile authentication default
  method 1 local
  method 2 none
  server-timeout 10

context ip router

  interface eth0
    ipaddress xxx.xxx.xxx.11 255.255.255.0
    mtu 1500

  interface eth1
    ipaddress 192.168.1.1 255.255.255.0
    mtu 1500

context ip router
  route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.193 0

context cs switch

  routing-table called-e164 SIP2ISDN
    route default none
    route 01234567 none
    route 012345670 none
    route 01234567[1-9] dest-table SIP2ISDN_EXT mapSIP2ISDN_To

  routing-table called-e164 ISDN2SIP
    route default dest-interface SIP_IF mapISDN2SIP_To

  routing-table called-e164 SIP2ISDN_EXT
    route 0 none
    route default dest-interface ISDN_PBX

  mapping-table called-e164 to called-uri mapISDN2SIP_To
    map default to sip:[EMAIL PROTECTED]

  mapping-table called-e164 to called-e164 mapSIP2I
    map default to 0
    map 01234567(.+) to \1

  interface isdn ISDN_PBX
    route call dest-table ISDN2SIP

  interface sip SIP_IF
    bind gateway sipgw
    route call dest-table SIP2ISDN

context cs switch
  no shutdown

gateway sip sipgw
  call-transfer-version 5
  session-timer-version 8
  bind interface eth0 router
  no shutdown
  use profile voip myvoip

port ethernet 0 0
  medium 10 half
  encapsulation ip
  bind interface eth0 router
  no shutdown

port ethernet 0 1
  medium 10 half
  encapsulation ip
  bind interface eth1 router
  shutdown

port bri 0 0
  clock auto
  encapsulation q921


q921 protocol pmp

    uni-side auto
      encapsulation cc-isdn
      bind interface ISDN_PBX switch

port bri 0 0
  no shutdown

port bri 0 1
  clock auto
  encapsulation q921

  q921
    protocol pmp
    uni-side auto
    encapsulation q931

    q931
      protocol dss1
      uni-side net
      encapsulation cc-isdn

port bri 0 1
  shutdown

----------------------------------------------







Michael Lyszczek wrote:
Any have any success using a patton smartnode 4118/js/eiu fxs gateway
with asterisk?  We we're able to get the unit to register with
asterisk, but when trying to place a call, no codec was compatible,
even though I had all of the following enabled on the patton ...


# G.711 A-Law/µ-Law (64kbps) # G.726 (ADPCM 40, 32, 24, 16 kpbs) # G.723.1 (5.3 or 6.3 kbps) # G.729ab (8kbps)

the link to this product is :

http://commerce.patton.com/pe_products.asp?category=51&MiDAS_SessionID=e41363efa86e409caf79ab1fd9b32e49

?

thanks for any help,
Michael Lyszczek
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