[Serusers] Fwd: AS5300/E1 + SER -> PB : Cisco redirect calls on SER

[Stephen Kingham]

Alcée wrote:
> 
>>
>> Hello,
>> I try to use cisco AS5300 with SER, like this  :
>>     phones -> SER -> AS5300 -> PSTN
>> or
>>     phones -> AS5300 -> PSTN
>>                 |
>>                      \/
>>                   SER
>> But the second form does not work.


You do not have a dial-peer telling the AS5300 to route calls to the PSTN

dial-peer voice 1 pots
  destination-pattern .
  progress_ind alert enable 8
  direct-inward-dial
  port 0:D

and/or

dial-peer voice 10 pots
  destination-pattern 0.
  progress_ind alert enable 8
  direct-inward-dial
  prefix 0
  port 0:D


Also I would use SIP Server in Proxy mode, and
I would create an access list on both ethernet interfaces to only allow 
TCP and UDP port 5060 from your SIP Proxy.


>>
>>
>> I don't want people to directly ask for AS5300. So I follow the 
>> "Howto" on SER web page.
>> But nothing arrive on SER when I make a call directly to AS5300, The 
>> call is forward to pstn.
>> So I make this configuration to minimize the problem.
>> But nothing at all.
>> The "debug ccsip error" tells "no route to destination".
>>
>> Have you any idea ?
>>
>> thanks :)
>>
>> Alcée
>>
>> ! Last configuration change at 14:17:05 UTC Mon Apr 11 2005 by admin
>> ! NVRAM config last updated at 12:57:01 UTC Mon Apr 11 2005 by admin
>> !
>> version 12.3
>> service timestamps debug uptime
>> service timestamps log uptime
>> no service password-encryption
>> !
>> hostname VoIP
>> !
>> boot-start-marker
>> boot-end-marker
>> !
>> !
>> !
>> resource-pool disable
>> !
>> aaa new-model
>> !
>> !
>> aaa authentication login h323 group radius
>> aaa authorization exec h323 group radius
>> aaa accounting delay-start
>> aaa accounting connection h323 start-stop group radius
>> aaa nas port voip
>> aaa session-id common
>> ip subnet-zero
>> !
>> !
>> isdn switch-type primary-net5
>> !
>> voice rtp send-recv
>> voice echo-canceller extended
>> !
>> voice class codec 1
>>  codec preference 1 g729r8 bytes 40
>>  codec preference 2 g711ulaw
>> !
>> voice class codec 2
>>  codec preference 1 g723ar63 bytes 48
>>  codec preference 2 g723r63 bytes 48
>>  codec preference 3 g729br8 bytes 40
>>  codec preference 4 g729r8 bytes 40
>>  codec preference 5 g723ar53 bytes 40
>>  codec preference 6 g723r53 bytes 40
>>  codec preference 7 g711ulaw
>> !
>> fax interface-type modem
>> !
>> !
>> controller E1 0
>>  clock source line primary
>>  pri-group timeslots 1-31
>> !
>> controller E1 1
>>  pri-group timeslots 1-31
>> !
>> controller E1 2
>>  clock source line secondary 4
>>  pri-group timeslots 1-31
>> !
>> controller E1 3
>>  pri-group timeslots 1-31
>> gw-accounting syslog
>> gw-accounting aaa
>>  attribute h323-remote-id resolved
>>  acct-template callhistory-detail
>>  suppress pots
>> !
>> !
>> !
>> interface Ethernet0
>>  ip address 192.168.0.129 255.255.255.0
>> !
>> interface Serial0:15
>>  no ip address
>>  isdn switch-type primary-net5
>>  isdn protocol-emulate network
>>  no cdp enable
>> !
>> interface Serial1:15
>>  no ip address
>>  isdn switch-type primary-net5
>>  no cdp enable
>> !
>> interface Serial2:15
>>  no ip address
>>  isdn switch-type primary-net5
>>  no cdp enable
>> !
>> interface Serial3:15
>>  no ip address
>>  isdn switch-type primary-net5
>>  no cdp enable
>> !
>> interface FastEthernet0
>>  ip address 172.19.20.2 255.255.255.0
>>  duplex full
>>  speed 100
>> !
>> interface Dialer0
>>  no ip address
>> !
>> ip classless
>> ip route 0.0.0.0 0.0.0.0 172.19.20.1
>> no ip http server
>> !
>> !
>> logging facility local0
>> logging 192.168.0.2
>> !
>> !
>> radius-server host 192.168.0.2 auth-port 1812 acct-port 1813
>> radius-server key test
>> radius-server vsa send accounting
>> radius-server vsa send authentication
>> !
>> voice-port 0:D
>> !
>> voice-port 1:D
>> !
>> voice-port 2:D
>> !
>> voice-port 3:D
>> !
>> !
>> !
>> dial-peer voice 3777 voip
>>  huntstop
>>  preference 2
>>  application session
>>  destination-pattern 3777#T
>>  progress_ind setup enable 3
>>  progress_ind alert enable 8
>>  voice-class codec 2
>>  session protocol sipv2
>> session target sip-server
>> dtmf-relay cisco-rtp h245-signal h245-alphanumeric
>>  fax rate 9600
>>  icpif 50
>> !
>> !
>> sip-ua
>>  retry invite 3
>>  retry response 3
>>  retry bye 3
>>  retry cancel 3
>>  timers expires 300000
>>  timers connect 300
>>  sip-server ipv4:192.168.0.1
>> !
>> !
>> line con 0
>> line aux 0
>> line vty 0 4
>>  password intermobile
>> !
>> ntp server 192.168.0.2
>> end
> 
> 
> 
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-- 
Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer
mailto:Stephen.Kingham at aarnet.edu.au
Telephone +61 2 6222 3575 (office)
           +61 419 417 471 (mobile)
sip:Stephen.Kingham at aarnet.edu.au

Voice and Video over IP
for The Australian Academic Research Network (AARNet)
http://www.aarnet.edu.au