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

[Alcée]

I forgot to give this dial-peer.
Certainly because I would like when phones call directly the AS5300, 
that the signaling to be redirect to SER before establishing the call.



Le 18 avr. 05, à 09:49, Stephen Kingham a écrit :

>
> 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
>