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<DIV dir=ltr align=left><SPAN class=625064111-12052005><FONT face=Arial
color=#0000ff size=2>This has been great !! Thx Barney</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of
</B>barney<BR><B>Sent:</B> Jueves, 12 de Mayo de 2005 03:30 a.m.<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT size=2><FONT size=3>Hi,<BR><BR>I used C3640, but It was changed,
because of few DSP in it. However, <BR>configuration is same. It also depends
on used IOS version. Here are <BR>fragments from
configurations:<BR><BR>AS5300:<BR><BR>!<BR>clock timezone GMT
0
; in some Docs = necessary<BR>!<BR>isdn switch-type
primary-net5 ; I`m in Europe :-)<BR>isdn voice-call-failure
0<BR>!<BR>!<BR>voice call send-alert<BR>voice rtp send-recv<BR>!<BR>voice
service voip<BR>!<BR>voice class codec 3<BR> codec preference 1
g711alaw<BR> codec preference 2 g711ulaw<BR>!<BR>controller E1
0<BR> clock source line primary<BR> pri-group timeslots
1-31<BR> description to-PSTN<BR>!<BR>translation-rule
2
; type of number <BR>(subs/national/international) depend on your telco
provider<BR> Rule 0 .... 021111 ANY subscriber<BR> Rule 10 any
0211111111 ANY subscriber<BR>!<BR>!<BR>translation-rule
10
; type of number <BR>(subs/national/international) depend on your telco
provider<BR> Rule 0 ^421211110... 0 ANY subscriber<BR> Rule 1
^421211111... 1 ANY subscriber<BR> Rule 2 ^421211112... 2 ANY
subscriber<BR> Rule 3 ^421211113... 3 ANY subscriber<BR> Rule 4
^421211114... 4 ANY subscriber<BR> Rule 5 ^421211115... 5 ANY
subscriber<BR> Rule 6 ^421211116... 6 ANY subscriber<BR> Rule 7
^421211117... 7 ANY subscriber<BR> Rule 8 ^421211118... 8 ANY
subscriber<BR> Rule 9 ^421211119... 9 ANY subscriber<BR> Rule 10 any
1234 ANY subscriber<BR>!<BR>interface Serial0:15<BR> description
PRI-D-CHANNEL-to-PSTN<BR> no ip address<BR> no logging event
link-status<BR> isdn switch-type primary-net5<BR> isdn guard-timer
3000<BR> isdn map address 0.* plan isdn type subscriber<BR> isdn
send-alerting<BR> isdn sending-complete<BR> no cdp
enable<BR>!<BR>voice-port 0:D<BR> input gain -6<BR> output
attenuation 14<BR> echo-cancel coverage 32<BR> echo-cancel
suppressor<BR> cptone SK<BR> description E1<BR> bearer-cap
Speech<BR>!<BR>dial-peer voice 8 pots<BR> tone ringback
alert-no-PI<BR> destination-pattern 00T<BR> port 0:D<BR> prefix
00<BR>!<BR>dial-peer voice 10 pots<BR> tone ringback
alert-no-PI<BR> destination-pattern 0[1-9]........<BR> port
0:D<BR> prefix 00421<BR>!<BR>dial-peer voice 20 pots<BR> tone
ringback alert-no-PI<BR> destination-pattern
00421[1-9]........<BR> port 0:D<BR> prefix 00421<BR>!<BR>dial-peer
voice 999 voip<BR> numbering-type international<BR> incoming
called-number .<BR> voice-class codec 3<BR> session protocol
sipv2<BR> dtmf-relay cisco-rtp h245-signal h245-alphanumeric<BR> fax
rate 7200<BR> ip qos dscp cs5 media<BR> no
vad<BR> supplementary-service pass-through<BR>!<BR>dial-peer voice 1
pots<BR> incoming called-number
.<BR> direct-inward-dial<BR> port 0:D<BR>!<BR>dial-peer voice
42121111 voip<BR> destination-pattern
42121111....<BR> translate-outgoing called 10<BR> voice-class codec
3<BR> session protocol sipv2<BR> session target
ipv4:1.2.3.4:5060 ; IP address of Asterisk<BR> ip qos
dscp cs5 media<BR> no vad<BR>!<BR>sip-ua<BR> retry invite
3<BR> retry response 3<BR> retry bye 3<BR> retry cancel
3<BR> timers trying 1000<BR> sip-server
ipv4:1.2.3.4:5060 ; IP address of Asterisk<BR>!<BR>ntp
server 1.2.3.5<BR>!<BR><BR><BR>I`m not sure, if all things are necessary and
correct, but... it`s working <BR>:-). I can place calls from asterisk to PSTN
via AS5300, and also receive <BR>calls from pstn. In this configuration, i
have DDI prefix from my telco as <BR>42121111xxxx. 421 = international
prefix 2 (02) = national prefix, 1111xxxx <BR>is my DDI prefix in which i can
use 10 000 numbers.<BR><BR>I`m using 4 digit extensions in my numbering plan
at Asterisk, so I could <BR>have DID in 1:1 mapping.<BR><BR>Fragments of very
simple asterisk
configurations:<BR><BR>Extensions.conf<BR><BR>[globals]<BR>CISCOSIPGW=2.2.2.2
;(IP address of AS5300)<BR><BR>[outgoing-cisco-pstn]<BR>exten =>
_90NXXXXXXXX,1,Dial(SIP/${EXTEN:1}@${CISCOSIPGW},180) ; local
calls<BR><BR><BR>Sip.conf<BR><BR>[2.2.2.2]<BR>type=friend<BR>host=2.2.2.2<BR>nat=no<BR>canreinvite=yes<BR>dtmfmode=rfc2833<BR>disallow=all<BR>allow=alaw<BR>allow=ulaw<BR><BR>In
this cas, only 10 digit numbers are allowed (only national calls) to dial
<BR>via Cisco, through number 9 as an prefix for outbound calls.<BR><BR>Hope,
that this samples will be usefull for you.<BR><BR><BR>PS: sorry for english, i
hope, you could understand it :-)<BR><BR>-b<BR><BR><BR><BR><BR>----- Original
Message ----- <BR>From: "Anton Krall" <</FONT><A href=""><FONT
size=3>akrall-lists@intruder.com.mx</FONT></A><FONT size=3>><BR>To:
<</FONT><A href=""><FONT size=3>barney@spirit.bentel.sk</FONT></A><FONT
size=3>><BR>Sent: Wednesday, May 11, 2005 7:08 PM<BR>Subject: RE:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600<BR><BR><BR>> Hey
Barney<BR>><BR>> What are the steps necessary to make that work on the
cisco AS5300? Any<BR>> configs I need to check to make it work? And what do
I need on asterisks<BR>> side?<BR>><BR>> Ever used cisco
3600?<BR>><BR>> |-----Original Message-----<BR>> |From: </FONT><A
href=""><FONT
size=3>asterisk-users-bounces@lists.digium.com</FONT></A><BR><FONT size=3>>
|[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of barney<BR>>
|Sent: Miércoles, 11 de Mayo de 2005 05:22 a.m.<BR>> |To: Asterisk Users
Mailing List - Non-Commercial Discussion<BR>> |Subject: Re:
[Asterisk-Users] Asterisk and Cisco AS5300 or 3600<BR>> |<BR>> |>
Just in case you don't know, AS5350 supports SIP *and* H323<BR>> |after
IOS<BR>> |> version<BR>> |> 12.3 (maybe a little earlier).<BR>>
|> It allows you to use both at the same time, without needing<BR>> |to
set it<BR>> |> up for one system specifically.<BR>> |> Haven't
tried it with Asterisk yet though.<BR>> |<BR>> |<BR>> |I have tried
it. I have SIP trunk between Asterisk and AS5300<BR>> |(C3640 before), and
it`s working good.<BR>> |It`s quite good solution, but its much more
expensive as some<BR>> |PCI card direct in Asterisk (i`m using PRI
interconnect to PSTN).<BR>> |<BR>> |-b<BR>> |<BR>> |PS: sorry for
poor english<BR>> |<BR>> |<BR>> |<BR>> |> On Wednesday 11 May
2005 11:23, Anton Krall wrote:<BR>> |>> I need some advice on some
h323 issues. I need to test connectivity<BR>> |>> from Asterisk to a
Cisco AS5300 that has PSTN lines and to<BR>> |cisco 3600<BR>> |>>
voip routers.<BR>> |>><BR>> |>> H323 needs to be used here
but I was wondering if anybody<BR>> |has linked<BR>> |>> Asterisk
to these Cisco routers before?<BR>> |> Just in case you don't know,
AS5350 supports SIP *and* H323<BR>> |after IOS<BR>> |>
version<BR>> |> 12.3 (maybe a little earlier).<BR>> |> It allows
you to use both at the same time, without needing<BR>> |to set it<BR>>
|> up for one system specifically.<BR>> |> Haven't tried it with
Asterisk yet though.<BR>> |><BR>> |> Richard.<BR>> |>
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