[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk

[Pavlidis Savas]

I have a peculiar problem.
I have installed asterisk
and also g729 (2 channels).
I have a Cisco7940 IP phone
with SIP installed (v6)
and a cisco router 2650xm
which has an isdn bri voice
interface that connects to
a legacy pbx system. Also
I installed a x-lite
to make some tests.

I have configured everything
after a lot of search and
trial and error. So I have
managed to make calls from the
7940 to x-lite and vice-versa
and also to make calls to
to legacy phones from the
7940 or the x-lite via the
cisco router using its voice
interface.
BUT the problem is that from
the legacy PBX phones I can call
the x-lite but not the cisco
7940 IP Phone.
Where is the problem????
Can anyone help me?

here are the configurations:
SIP.CONF
[general]
context=default                 ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port 
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds 
to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g729


[xlite1]
type=friend
regexten=1239                 ; When they register, create extension 1239
username=xlite1
callerid="Savas Pavlidis" <1239>
host=dynamic
;nat=yes                       ; X-Lite is behind a NAT router
canreinvite=no                ; Typically set to NO if behind NAT

[10.1.1.1]                      ; Cisco 2650XM router
type=friend
host=10.1.1.1
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=g729

[419]                           ; 7940G Cisco IP Phone
type=friend
username=419
host=dynamic
canreinvite=yes
dtmfmode=inband
disallow=all
allow=g729


EXTENSIONS.CONF (PART OF IT)
; The numbers 3XX belong to the traditional
; PBX telephones.
;
exten => _3XX,1,Dial(SIP/${EXTEN}@10.1.1.1)
exten => _3XX,n,Congestion

;
;
;
exten => 419,1,Dial(SIP/419)

exten => 420,1,Dial(SIP/xlite1)
exten => 420,2,Congestion


; as you may understand 419 is the cisco ip phone
; and extension 420 is the softp phone x-lite
; on the pc.


CISCO ROUTER CONFIGURATION (PART OF IT)
dial-peer voice 1 pots
 destination-pattern 3..
 direct-inward-dial
 port 1/0/0
 forward-digits all
!
dial-peer voice 2 pots
 destination-pattern 3..
 direct-inward-dial
 port 1/0/1
 forward-digits all
!
!
dial-peer voice 100 voip
 destination-pattern 9..
 session target ipv4:100.0.0.1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 101 voip
 destination-pattern 8..
 session target ipv4:100.0.0.1
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 103 voip
 destination-pattern 1..
 session target ipv4:200.200.201.2
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 200 voip
 destination-pattern 40.
 session target ipv4:100.0.0.191
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 201 voip
 destination-pattern 5..
 session target ipv4:100.0.0.191
 dtmf-relay h245-signal h245-alphanumeric
 ip qos dscp cs5 media
!
dial-peer voice 202 voip
 destination-pattern 42.
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 205 voip
 destination-pattern 41.
 session protocol sipv2
 session target sip-server
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:10.1.1.250:5060
!

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