similar to: UNISTIM channel driver available

I've been trying to look up information on upgrading firmware on a nortel i2004 ip phone. I have this phone leftover from a trial, and it's supposed to be upgradable to current firmwares. Since I also run a DMS I was able to login to nortel's site and get all the firmware files, but All the NTP's regarding firmware upgrading these are how to tell you BCM to send the file to it.

Hi, Does anybody have an updated Chan Unistim that compiles on Asterisk 1.2beta? Below is the output when compiling on Red Hat 9.0 Thanks, [root@maui2 chan_unistim-0.9.2]# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_unistim.o

Is anyone successfully using this phone with Asterisk? There is a lot mentioned about CISCO but nothing about Nortel... Alex.

Hello, I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on their asterisk implementation. My local dealer says they are not compatible with any open source implementations. Is there a phone compatibility list somewhere? Cheers Cian

I'm trying chan_unistim-1.0.0.5e with asterisk-1.4.22 and Nortel i2002 phone. When I dial the numbers are not showing up on a display, and the is no frequency sound when pressing the numbers. I think it is related to chan_unistim, isn't it. Did anybody encounter this problem and/or solve it? -- #Joseph GPG KeyID: ED0E1FB7

Anyone know of one that I could use? Thanks, Shri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/51669091/attachment.htm

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

I have in "unistim.conf [violet] ... context=internal but it is not recognized. When I try to make a call it looks for context "default" Is it a bug or a limitation of unistim. -- #Joseph GPG KeyID: ED0E1FB7

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

Anyone get the Nortel 1140E phones working with Asterisk ? These look like great phones and I would like to start using them on our deployments. I know these will work with Asterisk but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi All, During last three month I have worked on improving functionality of Nortel phones working with asterisk to replace existing Nortel station by asterisk. Many improvments done, listed below. I have only i2002 phone and unable to test if new version of channel correctly works with i2204 phone. If anyone can test it and report issues, it would be great. Please visit mantis to find out patch

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

Hello! I am semi-new to asterisk, I've been toying with it for about a month now. I'm using the current version from the CVS server (as of last week) and have succesfully connected it to Broadvoice, as well as a few softphones. I got a Nortel i2004 IP phone from a friend to setup at home as a homephone using Broadvoice, but going through my * box and using iptel.com for toll free calls.

I am trying to compile chan_unistim and getting some compilation errors: Asterisk Version: Asterisk CVS-03/21/05-12:24:27 AstHeaders: /usr/include Gcc: gcc version 3.2.3 20030502 (Red Hat Linux 3.2.3-42) OS: CentOS 3.3 (final) The isssue seems to be stemming from AST_MUTEX_DEFINE_STATIC - here is the error dump from chan_unistim: gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g

Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI> core show channeltypes I would have response like: connect*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- -----------

Hello, is it possible to connect a Nortel i2004, (i2002) IP Phone to an * PBX? I have seen on the net that there should be an SIP firmware aviable for the phone, anyone an idea where to get it? Because orginally they use a own protocol to connect to an Meridian, BCM or CSE1K. Another idea is to write a module for * that can handle this protocol. bye SLiME

Hi all, Anyone know why this happens? Mem: 524288k total, 508120k used, 16168k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856