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You will need to flash the SIP firmware to the phone in order to get it to work with FreePBX. If you have one of these phones, there is a good chance that it is running the old UNISTIM firmware. ![]() This guide is not intended for beginners. #Freepbx install unistim asterisk trialSetting these phones up with FreePBX/Astericks was a chore and after many hours of trial and error and google searching, i’ve managed to get these working pretty well with FreePBX. I had a few Avaya/Nortel 1120e/1140e IP phones laying around at home and decided to create a house phone and intercom system. #Freepbx install unistim asterisk seriesVoip, freepbx, asterisk, ip, phones, nortel, avaya, 1100, 1120e, 1140e, series For extra debug, uncomment #define DUMP_PACKET 1 and recompile chan_unistim.9 mins read Using Avaya 1100 Series IP phones with FreePBX If you have difficulties, try unistim debug and set verbose 3 on the asterisk CLI.3 can be used on black i2004 with chrome. 1 can be used on new firmware (black i2004) and 2 on old violet i2004. You can modify yourphone rtp_method= with 0, 1, 2 or 3. At this time, I know three ways to establish a RTP session. Don't forget : this work is based entirely on a reverse engineering, so you may encounter compatibility issues.If you don't do that or the bindaddr is invalid (or no longer valid, eg dynamic IP), phones should be able to display messages but will be unable to send/receive RTP packets (no sound) If asterisk is behind a NAT, you must set general public_ip= with your public IP.You can use IAX2 trunking if you're master asterisk is outside. Only one phone per public IP (multiple phones behind the same NAT don't work).If a phone is behind a NAT, you should port forward UDP 5000 (or change general port= in nf) and UDP 10000 (or change yourphone rtp_port=) Check if nf is in the correct directory.Set public_ip with an IP of your computer.Comment #define HAVE_IP_PKTINFO in chan_unistim.c.Use either E1/T1 trunks, or buy UTPS (UNISTIM Terminal Proxy Server) from Nortel. It's not possible to connect a Nortel Succession/Meridian/BCM to Asterisk via chan_unistim.RTP NAT - nat (control ast_rtp_setnat, default = 0. ![]() Language - language (see section Coutry Code).Call Group - callgroup pickupgroup (untested).This feature requires chan_local (loaded by default).callhistory=0 will NOT disable normal asterisk CDR logs. If history files were created, you also need to delete them. It can be a privacy issue, you can disable this feature by adding callhistory=0. By default, chan_unistim add any incoming and outgoing calls in files (/var/log/asterisk/unistimHistory).Use the two keys located in the middle of the Fixed feature keys row (on the bottom of the phone) to enter call history.If a user enter TN 1234, the phone will be known as USTM/.If you want to do that, add extension=line in your phone context. The line=> entry in nf does not add an extension in asterisk by default. NETMSK / DEF GW : netmask and default gateway.If you see "Locating server", power off or reboot the phone and try again.Press quickly the four buttons just below the LCD screen, in sequence from left to right.You can use a least a Nortel i2002, i2004 and i2050. #Freepbx install unistim asterisk driverThis is a channel driver for Unistim protocol. Unified Networks IP Stimulus (UNIStim) Channel Driver for Asterisk ![]()
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