Asterisk + libss7 [3rd]

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Connecting and placing calls:

[X ~]# cat /etc/asterisk/sip.conf

[general]
context=[your context] ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=[your IP] ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
tos_sip=cs3 ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
cos_sip=3
cos_audio=5

relaxdtmf=yes
canreinvite=no
disallow=all ; see doc/rtp-packetization for framing options
allow=g729 ; add whatever codec you have and like to use
;allow=ulaw
;allow=alaw
allowguest=no
notransfer=yes ; part of rtp timeout
rtptimeout=19
rtpholdtimeout=20

[client1]
type=peer
insecure=port,invite
host=[client ip]
notransfer=yes
disallow=all
allow=g729
;allow=g723
allow=ulaw
;nat=yes
qualify=no
canreinvite=no

[client2]
type=peer
insecure=port,invite
host=[client ip]
notransfer=yes
disallow=all
allow=g729
;allow=g723
allow=ulaw
;nat=yes
qualify=no
canreinvite=no


[x~]# cat /etc/asterisk/extensions.conf

[your context]

; Dial plan for client1
exten => _110XXXXXX.,n,Dial(DAHDI/g1/${EXTEN:2},80,L(3300000))
exten => _110XXXXXX.,n,Hangup

; End of 1st E1

; Dial plan for client2

exten => _220XXXXXX.,n,Dial(DAHDI/g1/${EXTEN:2},80,L(3300000))
exten => _220XXXXXX.,n,Hangup





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Those are my general setting .

Have done quit a lot change in production setting . If you face any issue please drop me a line . I will try to solve that .


thanks
Salaque

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