[Asterisk-Users] Cisco AS5400 Configuration as a SIP Peer - URGENT

[Matt Roth]

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