Nortel 1165e IP Phone. Nortel has released UNIStim firmware 4.0 for their IP phones; 0621C7A for IP Phone 2007, 0623C7F, 0624C7F, 0625C7F and 0627C7F for IP Phone. Posted on April 7. We sell the E-metrotel UCX server that will do all of this and you can use any of your Nortel Unistim phones as well as digital phones.

Keeping the Nortel CS1000 Alive Avaya's announcement of a 'final release' of Nortel CS1000 at version R7.6 appears to have left these customers with one choice: rip and replace. Avaya's announcement of a 'final release' of Nortel CS1000 at version R7.6 appears to have left these customers with one choice: rip and replace.

Enterprises that have Nortel CS1000 platforms and phones were hit hard by the Nortel's bankruptcy. When Nortel's telecom related assets were acquired by Avaya and GENBAND, the CS1000 customers became dependent on Avaya for ongoing upgrades and maintenance. Avaya's announcement of a 'final release' of CS1000 at version R7.6 appears to have left these customers with one choice: rip and replace. Rip and Replace The rip and replace solution presents several problems for the CS1000 owner. First, there is the expense of a new platform; in most cases new servers will be required. There certainly will be new software, management tools, operational procedures, and re-training of IT staff. Beyond the core platform, there is also their considerable investment in desk phones.

Many enterprises have proprietary Nortel TDM and/or Nortel UNIStim IP phones that are not supported by other vendors. With nearly half the cost of an upgrade tied to desk phones, replacing endpoints can be one of the biggest costs, and, of course, every new phone requires an end user to be retrained. And, since CS1000 customers have felt the pain of being tied to one vendor, standards-based SIP endpoints are in-demand. The technicians that support the CS1000 and UNIStim phones are also moving on: either retiring or learning new systems. As CS1000 technicians refocus on other platforms, their numbers dwindle, impacting availability. As time goes by, the shortage in expertise and parts drives up maintenance fees. Users will have to be re-trained.

New apps will need to be introduced that essentially replace the CS1000 apps. UNIStim phones will have to be replaced. Moving to Unified Communications will be limited unless the CS1000 is eliminated. IT's Goals Given the expense and complexity of rip and replace it's no surprise that this is not a CS1000 owner's first choice. An IT team's goals for the continued life of the CS1000 are likely to include: • Extending the life of legacy UNIStim endpoints that can't be converted to SIP • Building a true SIP-based solution • Maintaining feature transparency, thereby saving staff time and retraining users • Using the enterprise network or Cloud-based service • Adding Mobility (smart devices) • Offering UC to a range of endpoints (PC/Mac, tablet, phone) • Adding Video & Collaboration • Supporting Advanced Messaging A GENBAND Solution GENBAND has acquired Nortel's carrier assets.

Using their acquired intellectual property they for the CS1000 owner. All three pathways extend the useful life of the CS1000. The pathways not only allow the retention of the CS1000 investment, they also permit the addition of new capabilities for the users. Three GENBAND Solutions Wrap the CS1000 with a Cloud Overlay This solution uses the GENBAND cloud-based service called NUViA™. NUViA™ is an enterprise-class Unified Communications as a Service designed to eliminate the need for premises-based session/call control. It is powered by GENBAND's EXPERiUS™ solutions which is a platform that ties its heritage to the Nortel MCS platform.

NUViA services can overlay the CS1000 implementation without replacing the existing CS1000 hardware or software. It offers an overlay of UC, video and mobility applications on top of the CS 1000. The enterprise is free to use as little or as much as they want since it is sold on a per seat basis. Migrate Endpoints to Cloud Based Core This path expands the NUViA cloud based solution. Crack Project Igi 2 Covert Strike Password Keeper. The enterprise re-registers the UNIStim IP or SIP phones into the cloud-based NUViA system.

The DID's can be moved into SIP or cloud connections or can be maintained on premise gateways registered into NUViA. This allows the enterprise to be always current with the latest features. This solution is also priced on a per seat basis. Migrate Endpoints to Premises Based Core This third alternative moves the session/call control to GENBAND's EXPERiUS™ service.

EXPERiUS ties its heritage to the Nortel MCS application server. Given the heritage the feature set will be familiar to CS1000 users.

However EXPERiUS is very much a fully virtualized platform with a hardware freedom model where enterprises can select Dell, HP, or IBM servers. The enterprise would re-register UNIStim, IP or SIP phones to the on-premises EXPERiUS servers. Enterprises can then add UC, video and mobility applications. GENBAND's Intelligent Messaging Manager integrates with EXPERiUS for voice mail, including emulation of much of the Call Pilot's telephone interface. The Benefits The benefits of the three alternative solutions are: • Limited or no user training except for new features • The option to select either CAPEX solution with on-premises or an OPEX solution via the cloud • Retaining the Nortel phones can save $200 to $500 per user • Migrating forward with a SIP-based solution • Deploying a full set of UC features • No SIP device licenses Conclusions The enterprise may select to rip and replace their IP PBX. However, the GENBAND solutions offer a lower impact path for the future. The financial expenses will be lower if the enterprise can continue using its existing CS1000 and UNIStim phone investment.

This is not to say that the GENBAND is the best solution. The enterprise should look at the ROI on both the GENBAND solutions and a rip and replace implementation. It may be that the enterprise is ready to replace the UNIStim phones and the CS1000 is at the end of its ROI life. The GENBAND solutions offer another benchmark for comparison.

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This is a channel driver for the UNISTIM (Unified Networks IP Stimulus) protocol. It provides UNISTIM server services that you can use to drive Nortel i2002, i2004, i2007 and i2050 phones. The following features are supported: • Transfer • Call Forward • Message Waiting Indication (MWI) • Distinctive ring • Call History • Send/Receive CallerID • Redial • (Dynamic) SoftKeys • SendText() • Music On Hold Available on all version of Asterisk. With described fature set available in Asterisk 11.

For 1.8 and 10 can be downloaded as patches. How to place a call: The line=>entry in unistim.conf does not add an extension in asterisk by default. If you want to do that, add extension=line in your phone context. If you have this entry on unistim.conf: [violet] device=006038abcdef line =>102 then use exten =>2100,1,Dial(USTM/102@violet) You can display a text with: exten =>555,1,SendText(Sends text to client. Greetings) Distinctive ring: You need to append /r to the dial string.

The first digit must be from 0 to 7 (inclusive). It’s the ‘melody’ selection. The second digit (optional) must be from 0 to 3 (inclusive). It’s the ring volume. 0 still produce a sound. Select the ring style #1 and the default volume: exten =>2100,1,Dial(USTM/102@violet/r1) Select the ring style #4 with a very loud volume: exten =>2100,1,Dial(USTM/102@violet/r43) Country code: You can use the following codes for country= us fr au nl uk fi es jp no at nz tw cl se be sg il br hu lt pl za pt ee mx in de ch dk cn If you want a correct ring, busy and congestion tone, you also need a valid entry in indications.conf and check if res_indications.so is loaded.

Language= is also supported but it’s only used by Asterisk (for more informations see ). The end user interface of the phone will stay in english.

Bookmarks, Softkeys: – Layout: ——————– 5 2 4 1 3 0 – When the second letter of bookmark= is @, then the first character is used for positioning this entry – If this option is omitted, the bookmark will be added to the next available sofkey – Also work for linelabel (example: linelabel=”5@Line 123″) – You can change a softkey programmatically (since 1.0.0.4) with SendText(@position@icon@label@extension) ex: SendText(@1@55@Stop Forwd@908) Autoprovisioning This feature must only be used on a trusted network. It’s very insecure: all unistim phones will be able to use your asterisk pbx. Autoprovisioning=yes You must add an entry called [template]. Each new phones will be based on this profile. You must set a least line=. This value will be incremented when a new phone is registred.

Device= must not be specified. By default, the phone will asks for a number. It will be added into the dialplan.

Add extension=line for using the generated line number instead. Example: [general] port=5000 autoprovisioning=yes [template] line =>100 bookmark=Support@123; Every phone will have a softkey Support If a first phone have a mac = 006038abcdef, a new device named USTM/100@006038abcdef will be created. If a second phone have a mac = 00, it will be named USTM/100000 and so on.

Autoprovisioning=tn In this mode, new phones will ask for a tn, if this number match a tn= entry in a device, this phone will be mapped into. Example: [black] tn=1234 line =>100 If a user enter TN 1234, the phone will be known as USTM/100@black. History: – Use the two keys located in the middle of the Fixed feature keys row (on the bottom of the phone) to enter call history. – By default, chan_unistim add any incoming and outgoing calls in files (/var/log/asterisk/unistimHistory). It can be a privacy issue, you can disable this feature by adding callhistory=0. If history files were created, you also need to delete them.

Callhistory=0 will NOT disable normal asterisk CDR logs. Forward: – This feature requires chan_local (loaded by default) Generic asterisk features: You can use the following entries in unistim.conf – Billing: accountcode amaflags – Call Group: callgroup pickupgroup – Music On Hold: musiconhold – Language: language (see section Coutry Code) – RTP NAT: nat (control ast_rtp_setnat, default = 0. Obscure behaviour) Trunking: It’s not possible to connect a Nortel Succession/Meridian/BCM to Asterisk via chan_unistim. Use either E1/T1 trunks, or buy UTPS (UNISTIM Terminal Proxy Server) from Nortel. Issues: • As always, NAT can be tricky. If a phone is behind a NAT, you should port forward UDP 5000 (or change [general] port= in unistim.conf) and UDP 10000 (or change [yourphone] rtp_port=) • Only one phone per public IP (multiple phones behind the same NAT don’t work).

Setup a VPN if you want to do that. • If asterisk is behind a NAT, you must set [general] bindaddr= (0.9.2) or public_ip (0.9.4) with your public IP. 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) • Don’t forget: this work is based entirely on a reverse engineering, so you may encounter compatibility issues. At this time, I know three ways to establish a RTP session. You can modify [yourphone] rtp_method= with 0, 1, 2 or 3. 0 is the default method, should work. 1 can be used on new firmware (black i2004) and 2 on old violet i2004.

3 can be used on black i2004 with chrome. • If you have difficulties, try unistim debug and set verbose 3 on the asterisk CLI. For extra debug, uncomment #define DUMP_PACKET 1 and recompile chan_unistim. I have a new problem, however: Using this new version of chan_unistim, my Nortel phones are malfunctioning 🙁 When I call them from another phone, they start sending RTP packets IMMEDIATELY even before picking up the receiver. This doesn’t cause problem if Ipick up the receiver, speak with the caller and then put it down, because in this case the RTP stream stop after the conversation. However, if I don’t pick up the receiver of the Nortel phone, then it keeps sending the RTP packets indefinitely, even after Asterisk informs it that the caller has put down the receiver. So, the Nortel phone remains in such a state that it keeps bombing the Asterisk server with RTP packet, and when I call it next time or I initiate a call from it, then unistim signaling still seems to work but there is no audio 🙁 This is a huge problem, since an unanswered call on a Nortel phone renders it useless.

This problem didn’t appear in earlier Asterisk versions (e.g. 1.8 or even 10), but those versions didn’t handle the dialplan and sharpdial or interdigit_timer options as I described previously. What do you think? Is it a feature or a bug? And if it’s a bug, where can i report it? Thank you in advance.

Dear Igor, I tried to contact you about two years ago, but no reply. Chan unistim as compiled on 11.20 (11.23 starts to break because ooh323 doesn’t connect to legacy phone systems, whereas the discontinued h323 does, so we need the earlier 11.20) on a raspberry pi, and banana, and orange, never closes UDP ports.

I presumed this was an issue with the unknown cpu type, but no. I’ve compiled it correctly with all modern dependencies on a genuine intel P4 system Same problem. Every call, or operation involving RTP ports leaves them stuck indefinitely on the host system. I have a manually-set narrow udp range, but the asterisk default of nearly 65543 ports probably means most, no else sees this behaviour between reboots. I did a force into the current stable 11.23 as from Digium to make chan_h323 compile, thinking chan_unistim would be fixed, but it’s not. It still leaves udp ports open, with data sitting in their rx or tx buffers.

Is there a way to fix this? Or why is this major bug still in the accepted current long term stability build from Digium? I’d be happy to modify the chan_unistim.c source if it’s a matter of some changes Thanks for your time. This is truly frustrating, as the Nortel sets are immensely better than any SIP set on the market. (Perhaps barring the aastra ones built for nortel?) •. Hi Igor, 1) I initially didn’t create one on asterisk.org as I was on a pi, and didn’t think it would be taken seriously. Now duplicated on x86 I will.

2) Using simply netstat -l (for listening) and the port range as defined in rtp.conf (I use 9 for example), calls grab a pair, then fill up, two by to. At 100 ports, it runs out soon, and nothing then gets a udp port. Eg: udp 4288 0 x.x:31028 *:* udp 0 0 x.x:31029 *:* udp 12864 0 x.x:31038 *:* udp 0 0 x.x:31039 *:* udp 4288 0 x.x:31072 *:* udp 0 0 x.x:31073 *:* udp 12864 0 x.x:31078 *:* udp 0 0 x.x:31079 *:* udp 4288 0 x.x:31090 *:* udp 0 0 x.x:31091 *:* I made 5 test calls.