[Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT
Leandro Tenorio
leandro_tenorio at ciudad.com.ar
Tue Sep 13 17:54:38 MST 2005
K, I'll make a page under wiki when I have my password back (I forgot it), I
saw a lot of msg like this one.
There are several ways to configure it, below is one for in/out.
BTW, In Cisco config's it's important to add security, to just let pass the
call from your asterisk, Qos, etc. (not included) anyway it will work with
this one.
Some of the config in the Cisco isnt needed to run but the
troubleshooting will be easier
Just another recommendation, try to use the latest IOS and Nextport
firmware with just the sw you need.
Sip.conf
[name]
Type=friend
Host=xxx.xxx.xxx.xxx
Insecure=very
;Codecs you want to use
Disallow=all
Allow=g729
Allow=ulaw
Allow=alaw
DTMFMode=rfc2833
Cisco config
!ISDN type
isdn switch-type primary-dms100
!if you have several E1s/T1s you could want to make a trunk group
trunk group OutTrunkGroup
!how many call you want out
max-calls voice 96 direction out
!hunt scheme
hunt-scheme least-used both up
!
voice service pots
!to use T38 when fax is detected
fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through
g711ulaw
!
voice service voip
!to use T38 when fax is detected
fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through
g711ulaw
h323
sip
!To define diferent codes in a priority order and group them
voice class codec 1
codec preference 1 g729r8
codec preference 2 g726r16
codec preference 3 g723r63
codec preference 4 g711u
codec preference 5 g711a
!E1/T1 settings
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
controller T1 1
!same as before
interface Serial0:23
no ip address
isdn switch-type primary-dms100
isdn incoming-voice modem
isdn send-alerting
isdn bchan-number-order ascending
isdn sending-complete
!Trunk group previously defined
trunk-group OutTrunkGroup
no cdp enable
!
interface Serial1:23
!same as before
voice-port 0:D
echo-cancel coverage xx
no comfort-noise
no vad
! Used for input gain
input (in db)
! Used for output attenuation
output (in db)
voice-port 1:D
! Same as before
!just if you use a GK in H323 environment and you need to register
interface Serial3:0 / ethernet0 / fastethernet0 / etc
ip address xxx.xxx.xxx.xxx
h323-gateway voip interface
h323-gateway voip id GKNAME ipaddr xxx.xxx.xxx.xxx 1719
h323-gateway voip h323-id GWH323ID
!Incomming Calls First T1
dial-peer voice 1 pots
preference 1
direct-inward-dial
port 0:D
!Incomming Calls Second T1
dial-peer voice 2 pots
preference 1
direct-inward-dial
port 1:D
dial-peer voice 3 pots
preference 1
!Here you must set every DID you want to get to asterisk
! (dots) are patterns to match
incoming called-number 1. or called-number 718.
!Outgoing Calls to Trunkgroup
dial-peer voice 4 pots
trunkgroup OutTrunkGroup
huntstop
preference 1
! Unless define the digits to be stripped/added, in pots dialpeers,
the GW will not fw the digits you explicit before the . dot (in this case
the GW will just send everything after 1). am I clear? Sorry for my english
! I strongly suggest to send to the GW some prefix and cut it here
just to be a little more secure.
destination-pattern 1.
dial-peer voice 5 voip
huntstop
!matched digits in the pots inbound peer
destination-pattern 718.
!codecs defined at the beggining
voice-class codec 1
session protocol sipv2
session target dns:dns-name-of-the-proxy or ipv4:xxx.xxx.xxx.xxx ipaddress
!rfc2833 DTMF
dtmf-relay rtp-nte
dial-peer voice 6 voip
huntstop
destination-pattern 718.
voice-class codec 1
!for H323
session target ras
dtmf-relay rtp-nte
dial-peer voice 7 voip
incoming called-number 1.
voice-class codec 1
session target dns:dns-name-of-the-gk or ipv4:xxx.xxx.xxx.xxx ipaddress
dtmf-relay rtp-nte
!
dial-peer voice 8 voip
!traffic from SIP proxy
incoming called-number 1.
voice-class codec 1
session protocol sipv2
session target dns:dns-name-of-the-proxy or ipv4:xxx.xxx.xxx.xxx ipaddress
dtmf-relay rtp-nte
Hope this helps,
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt Roth
Sent: Tuesday, September 13, 2005 6:43 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT
List users,
It's been a while since I've posted here, but I've been hard at work pushing
toward our large scale Asterisk goal and keeping up with this list can be a
full time job by itself (I have19,543 unread list messages!!).
This Friday, September 16th 2005, my team will be at the MCI Development Lab
in Richardson, Texas testing our setup. We have a three server system
consisting of a Dell PowerEdge 6850 running Asterisk with the
cdr_addon_mysql.so module, a Dell PowerEdge 1850 running AstManProxy and
MySQL (our reporting server), and another Dell PowerEdge 1850 running
software we developed for indexing and archiving our digital recordings.
Our test setup has a second Asterisk server with a Digium quad-span card in
it acting as a TDM-VoIP gateway. We are shooting for scalability, so the
Asterisk server itself does no transcoding or DSP.
We have noloaded all codecs except one and moved any of the
resource-intensive activities to the gateway and the support servers.
Our production setup will replace the Asterisk TDM-VoIP gateway with a Cisco
AS5400HPX Universal Gateway. MCI has an AS5400 waiting for us at the D-Lab,
and while they are familiar with most aspects of it, they lack any
experience configuring it as a SIP peer for Asterisk. If anyone has
experience with this, please share it with me. Copies of your configuration
files from the AS5400 and your Asterisk server would be appreciated, as well
as any pointers to web resources. I'm personally inexperienced with the
AS5400, so the more information you can provide the better. It is my fear
that we will spend too much time configuring the AS5400 and miss out on an
opportunity to push the limits of the scalability of our design.
Ultimately, any advances we make in scaling Asterisk will be shared with the
community.
Basic connectivity of the AS5400 is an initial goal, but we have a few DSP
voice features that we need to configure:
* G.168 Echo Cancellation
* Jitter Buffering
* Comfort Noise Generation
* Disabling VAD/RTP Silence Suppression
Any relevant configurations from our current setup are after my signature.
I'm sorry for the short notice (a conference call with MCI exposed the need
for this message yesterday) and I will greatly appreciate any help you can
offer.
Thank you,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
======================================================================
Portion of /etc/extensions.conf from the Asterisk Gateway
; Context for passing incoming calls from our T1s to the Asterisk Server
[incoming] exten => _X.,1,NoOp("Inbound call for "${EXTEN}" from
"${CALLERID}) exten => _X.,2,Dial(SIP/${EXTEN}@sip_server)
exten => _X.,3,Congestion
Portion of /etc/sip.conf from the Asterisk Gateway
; Sip peer for the Asterisk Server
[sip_server]
type=peer ; Only call to this proxy, don't receive
calls from it
host=192.168.51.122 ; The IP of the SIP server
canreinvite=no ; Force the audio stream to remain on
Asterisk
dtmfmode=rfc2833 ; Use the RFC 2833 method of out-of-band
DTMF
Portion of /etc/extensions.conf from the Asterisk Server
; Context for passing outgoing calls to the Asterisk Gateway exten =>
_9X.,1,NoOp("Outbound call for "${EXTEN}" from "${CALLERID}) exten =>
_9X.,2,Dial(SIP/${EXTEN:1}@sip_gateway,60,tr) ; * removes the 9 and routes
the call exten => _9X.,3,Congestion
Portion of /etc/sip.conf from the Asterisk Server
; Sip peer for the Asterisk Gateway
[sip_gateway]
type=peer ; Only call to this proxy, don't receive calls
from it
host=192.168.51.121 ; The IP of the SIP gateway
canreinvite=no ; Force the audio stream to remain on Asterisk
dtmfmode=rfc2833 ; Use the RFC 2833 method of out-of-band
DTMF
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