SIP in higher education

Hi,
below i put my know-how for ser/asterisk and cisco call manager integration.
hope someone uses.

on ser login as root and add an asterisk user;
serctl add asterisk asteriskpwd asteriskmail

in ser.cfg do the following changes-ofcourse according to your dialplans-;

if(method == "INVITE" | method=="BYE" | method=="NOTIFY" | method=="CANCEL" ) 
{

if (uri =~ 
"sip:6[3,6,7][0-9][0-9]@*"){

route(2);
break;
}

}



......

route[2]
{
rewritehostport("your_asterisk_ip_address:5060");

if (!t_relay()) {
sl_reply_error();
};

}


.......


in trixbox do the following according to your dialplans;



;----------sip.conf------------------------

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]


port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=ulaw
;allow=alaw

; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
;context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf 

;i added below lines for ser integration--------------------------
register 
=>asterisk:asterisk_password_in_ser@yourseripaddress/asterisk

autocreatepeer=no
canreinvite=no


AND NOW;

"before extensions.conf modifications in default context create h323 trunk to 
cisco call manager and sip trunk to ser like this;

h323 trunk to cisco call manager:i explained in following link:
http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=2865&forum=6
with one dfiference change context to ser.

i.e context=ser (formerly default)


add sip trunk for ser:
CLICK ADD A "NEW SIP TRUNK"

Outgoing Dial Rules(surely these are my extensions change it for yours)
62xx
Outgoing Settings

trunkname:mysertrunk

PEER details:
context=ser
host=yourseripaddress
insecure=very
secret=yourasterisk_password_in_ser
type=peer
username=asterisk



then the following will be created in extensions_additional.conf

NUMBERS AT H323 TRUNK:

[outrt-001-h323-gateway]
include => outrt-001-h323-gateway-custom
exten => _6[3-9]XX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten => _6[3-9]XX,n,Macro(outisbusy,)
exten => _8XXXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten => _8XXXX,n,Macro(outisbusy,)
exten => _[1-4]XXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten => _[1-4]XXX,n,Macro(outisbusy,)

; end of [outrt-001-h323 gateway]

NUMBERS AT SER:

[outrt-002-serroute]
include => outrt-002-serroute-custom
exten => _62xx,1,Macro(dialout-trunk,1,${EXTEN},,)
exten => _62xx,n,Macro(outisbusy,)

; end of [outrt-002-serroute]


COPY THOSE BOTH AND -maybe with alittle change or with no change however 
youlike- PAST AS FOLLOWING then restart asterisk and ENJOY!



------extensions.conf-------------------------

[ser]
include => outrt-001-h323 gateway-custom
include => outrt-002-ser_route-custom
include => outrt-001-h323gateway-custom

exten => _6[3-9]XX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten => _6[3-9]XX,n,Playback(is-curntly-unavail)
exten => _6[3-9]XX,n,hangup

exten => _8XXXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten => _8XXXX,n,Playback(is-curntly-unavail)
exten => _8XXXX,n,hangup

exten => _[1-4]XXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten => _[1-4]XXX,n,Playback(is-curntly-unavail)
exten => _[1-4]XXX,n,hangup

include => outrt-002-ser_route-custom
exten => _62xx,1,Macro(dialout-trunk,1,${EXTEN},,)
exten => _62xx,n,Playback(is-curntly-unavail)
exten => _62xx,n,hangup


http://www.ergenay.com