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/[EMAIL PROTECTED]) 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 !

begin:vcard
fn:Savas Pavlidis
n:Pavlidis;Savas
email;internet:[EMAIL PROTECTED]
tel;work:+30 2310 573300
tel;fax:+30 2310 752280
x-mozilla-html:FALSE
version:2.1
end:vcard

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