[Asterisk-Users] Asterisk and Cisco AS5300 or 3600

Thu May 12 01:30:09 MST 2005

Hi,

I used C3640, but It was changed, because of few DSP in it. However, 
configuration is same. It also depends on used IOS version. Here are 
fragments from configurations:

AS5300:

!
clock timezone GMT 0                ; in some Docs = necessary
!
isdn switch-type primary-net5    ; I`m in Europe :-)
isdn voice-call-failure 0
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
!
voice class codec 3
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
!
controller E1 0
 clock source line primary
 pri-group timeslots 1-31
 description to-PSTN
!
translation-rule 2                                    ; type of number 
(subs/national/international) depend on your telco provider
 Rule 0 .... 021111 ANY subscriber
 Rule 10 any 0211111111 ANY subscriber
!
!
translation-rule 10                                ; type of number 
(subs/national/international) depend on your telco provider
 Rule 0 ^421211110... 0 ANY subscriber
 Rule 1 ^421211111... 1 ANY subscriber
 Rule 2 ^421211112... 2 ANY subscriber
 Rule 3 ^421211113... 3 ANY subscriber
 Rule 4 ^421211114... 4 ANY subscriber
 Rule 5 ^421211115... 5 ANY subscriber
 Rule 6 ^421211116... 6 ANY subscriber
 Rule 7 ^421211117... 7 ANY subscriber
 Rule 8 ^421211118... 8 ANY subscriber
 Rule 9 ^421211119... 9 ANY subscriber
 Rule 10 any 1234 ANY subscriber
!
interface Serial0:15
 description PRI-D-CHANNEL-to-PSTN
 no ip address
 no logging event link-status
 isdn switch-type primary-net5
 isdn guard-timer 3000
 isdn map address 0.* plan isdn type subscriber
 isdn send-alerting
 isdn sending-complete
 no cdp enable
!
voice-port 0:D
 input gain -6
 output attenuation 14
 echo-cancel coverage 32
 echo-cancel suppressor
 cptone SK
 description E1
 bearer-cap Speech
!
dial-peer voice 8 pots
 tone ringback alert-no-PI
 destination-pattern 00T
 port 0:D
 prefix 00
!
dial-peer voice 10 pots
 tone ringback alert-no-PI
 destination-pattern 0[1-9]........
 port 0:D
 prefix 00421
!
dial-peer voice 20 pots
 tone ringback alert-no-PI
 destination-pattern 00421[1-9]........
 port 0:D
 prefix 00421
!
dial-peer voice 999 voip
 numbering-type international
 incoming called-number .
 voice-class codec 3
 session protocol sipv2
 dtmf-relay cisco-rtp h245-signal h245-alphanumeric
 fax rate 7200
 ip qos dscp cs5 media
 no vad
 supplementary-service pass-through
!
dial-peer voice 1 pots
 incoming called-number .
 direct-inward-dial
 port 0:D
!
dial-peer voice 42121111 voip
 destination-pattern 42121111....
 translate-outgoing called 10
 voice-class codec 3
 session protocol sipv2
 session target ipv4:1.2.3.4:5060    ; IP address of Asterisk
 ip qos dscp cs5 media
 no vad
!
sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:1.2.3.4:5060    ; IP address of Asterisk
!
ntp server 1.2.3.5
!

I`m not sure, if all things are necessary and correct, but... it`s working 
:-). I can place calls from asterisk to PSTN via AS5300, and also receive 
calls from pstn. In this configuration, i have DDI prefix  from my telco as 
42121111xxxx. 421 = international prefix 2 (02) = national prefix, 1111xxxx 
is my DDI prefix in which i can use 10 000 numbers.

I`m using 4 digit extensions in my numbering plan at Asterisk, so I could 
have DID in 1:1 mapping.

Fragments of very simple asterisk configurations:

Extensions.conf

[globals]
CISCOSIPGW=2.2.2.2 ;(IP address of AS5300)

[outgoing-cisco-pstn]
exten => _90NXXXXXXXX,1,Dial(SIP/${EXTEN:1}@${CISCOSIPGW},180) ; local calls

Sip.conf

[2.2.2.2]
type=friend
host=2.2.2.2
nat=no
canreinvite=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw

In this cas, only 10 digit numbers are allowed (only national calls) to dial 
via Cisco, through number 9 as an prefix for outbound calls.

Hope, that this samples will be usefull for you.

PS: sorry for english, i hope, you could understand it :-)

-b

----- Original Message ----- 
From: "Anton Krall" <akrall-lists at intruder.com.mx>
To: <barney at spirit.bentel.sk>
Sent: Wednesday, May 11, 2005 7:08 PM
Subject: RE: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600

> Hey Barney
>
> What are the steps necessary to make that work on the cisco AS5300? Any
> configs I need to check to make it work? And what do I need on asterisks
> side?
>
> Ever used cisco 3600?
>
> |-----Original Message-----
> |From: asterisk-users-bounces at lists.digium.com
> |[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of barney
> |Sent: Miércoles, 11 de Mayo de 2005 05:22 a.m.
> |To: Asterisk Users Mailing List - Non-Commercial Discussion
> |Subject: Re: [Asterisk-Users] Asterisk and Cisco AS5300 or 3600
> |
> |> Just in case you don't know, AS5350 supports SIP *and* H323
> |after IOS
> |> version
> |> 12.3 (maybe a little earlier).
> |> It allows you to use both at the same time, without needing
> |to set it
> |> up for one system specifically.
> |> Haven't tried it with Asterisk yet though.
> |
> |
> |I have tried it. I have SIP trunk between Asterisk and AS5300
> |(C3640 before), and it`s working good.
> |It`s quite good solution, but its much more expensive as some
> |PCI card direct in Asterisk (i`m using PRI interconnect to PSTN).
> |
> |-b
> |
> |PS: sorry for poor english
> |
> |
> |
> |> On Wednesday 11 May 2005 11:23, Anton Krall wrote:
> |>> I need some advice on some h323 issues. I need to test connectivity
> |>> from Asterisk to a Cisco AS5300 that has PSTN lines and to
> |cisco 3600
> |>> voip routers.
> |>>
> |>> H323 needs to be used here but I was wondering if anybody
> |has linked
> |>> Asterisk to these Cisco routers before?
> |> Just in case you don't know, AS5350 supports SIP *and* H323
> |after IOS
> |> version
> |> 12.3 (maybe a little earlier).
> |> It allows you to use both at the same time, without needing
> |to set it
> |> up for one system specifically.
> |> Haven't tried it with Asterisk yet though.
> |>
> |> Richard.
> |> _______________________________________________
> |> Asterisk-Users mailing list
> |> Asterisk-Users at lists.digium.com
> |> http://lists.digium.com/mailman/listinfo/asterisk-users
> |> To UNSUBSCRIBE or update options visit:
> |>   http://lists.digium.com/mailman/listinfo/asterisk-users
> |>
> |
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> |
>
> 
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