This has been great !! Thx Barney
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:[EMAIL PROTECTED],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" <[EMAIL PROTECTED]> To:
<[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] >
|[mailto:[EMAIL PROTECTED] 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@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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