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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