I'm trying to move my company's phone system from Cisco's CUCM to an asterisk solution.
I have setup a PBXinaflash server, and setup an account with www.vitelity.com. I was able to get it working. I could send/receive calls, and it connected to an IVR.
Next I bought a Cisco 2821 ISR with a PRI card, as a backup for our current VOIP gateway. I was was able to load our current configuration on it, and after plugging in the PRI line it worked with our Cisco CUCM.
After that I did a goggle search on how to reconfigure the 2821. I downloaded the Cisco documentation on it, and found some example on Google. After my first try I was able to make outgoing phone calls. I was able to call a IVR and get it to recognize the digits I was dialing. The other phones Called ID showed the correct caller ID for us. The problem was getting incoming calls to work. So far I have gotten it to make a clicking sound, a dial tone, all line busy, and ring without any real phones ringing.
My objective it to get our asterisk IVR to pickup all calls on any of our phone numbers. We have ~150 DIDs, but only 4 are really used for anything.
My question is has anyone got this type of setup working? Is it possible? Could you share you configurations with me?
I think the problem is either the config for the 2821 or the trunk config in asterisk.
[Cisco 2821 ISR ~ 10.1.10.252]
version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname asterisk ! boot-start-marker boot-end-marker ! logging message-counter syslog ! no aaa new-model clock timezone GMT -6 network-clock-participate wic 1 network-clock-select 1 T1 0/1/0 ! dot11 syslog ip source-route ! ip cef ! no ipv6 cef ! multilink bundle-name authenticated ! isdn switch-type primary-ni ! voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip redirect ip2ip signaling forward unconditional fax protocol cisco sip bind control source-interface GigabitEthernet0/1 bind media source-interface GigabitEthernet0/1 ! voice-card 0 dspfarm ! archive log config hidekeys ! controller T1 0/1/0 pri-group timeslots 1-24 ! interface GigabitEthernet0/0 no ip address shutdown duplex auto speed auto ! interface GigabitEthernet0/1 ip address 10.1.10.252 255.255.255.0 duplex auto speed auto ! interface Serial0/1/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice isdn calling-number 3167945555 isdn supp-service name calling no cdp enable ! ip forward-protocol nd ip route 0.0.0.0 0.0.0.0 10.1.10.253 no ip http server no ip http secure-server ! control-plane ! voice-port 0/1/0:23 ! no mgcp package-capability res-package no mgcp package-capability fxr-package no mgcp timer receive-rtcp no mgcp explicit hookstate ! dial-peer voice 1 pots numbering-type unknown destination-pattern . direct-inward-dial port 0/1/0:23 forward-digits all ! dial-peer voice 2 voip numbering-type unknown destination-pattern . session protocol sipv2 session target ipv4:10.1.10.7:5060 session transport udp dtmf-relay rtp-nte codec g711ulaw ! gateway timer receive-rtp 1200 ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.10.7 ! line con 0 transport output telnet line aux 0 transport output telnet line vty 0 4 login transport input telnet line vty 5 15 login transport input telnet ! scheduler allocate 20000 1000 end
[asterisk trunk ~ 10.1.10.7]
Outgoing Settings Trunk Name: cisco2821 PEER Details: type=friend qualify=yes nat=no insecure=very host=10.1.10.252 dtmfmode=rfc2833 disallow=all context=from-internal allow=ulaw&alaw Incoming Settings USER Context: from-internal USER Details: type=friend qualify=yes nat=no host=10.1.10.252 dtmfmode=rfc2833 disallow=all context=from-trunk canreinvite=no allow=ulaw&alaw