Symptoms: Affected items Dialed + Missed + Received calls will not be displayed

Cause: Adobe has upgraded their software that has caused conflicts with the flash plugin.

Work Around: Bring up the phone’s web interface. PREFERENCES- find “Use Flash Plugin” change this setting to OFF. This will display the call login HTML form on the home page.

Fix: This has been brought to the attention of Snom development who are working on fix for this.

Nobody can deny the popularity that Skype has gained in the last years. It has become the largest VoIP network in the world with more than 230 million registered users. Many companies are in need to integrate their PBXs with this large community. They would want to be able to call or receive calls from business partners or travelling employees’ using Skype. In many cases, it is the only alternative and it is also free.
Companies can use Skype as telephony service provider also. Skype is a service provider for worldwide call termination thru its “Skypeout” service and for routing pstn inbound calling to skype accounts thru “Skypein” service which can provide DIDs from many countries.

Skip2PBX with IP PBX

Skip2PBX with legacy PBX

How it works: 

In the figures above, the Skip2Pbx gateway is connected to the company’s PBX thru traditional TDM, pots or IP (sip trunk) interface. The latter option applies when the company’s PBX is SIP enabled.
The gateway converts the analog signal, T1/E1 or SIP signal and audio streams to Skype signal and proprietary codec and vice versa allowing two way communication between any Skype client and the PBX.
By design Skype clients only support one voice channel at a time but Skip2PBX can bond multiple Skype clients/channels making it possible receive/place multiple Skype calls simultaneously and the company needs to advertise only one Skype contact. When calling outbound, the PBXs destination number cannot be a Skype alphanumeric contact but Skip2Pbx gateway maintains a table that maps numbers to Skype contacts so Pbx users still can call Skype users just by dialing their code as defined in the gateway. Since the transmission of voice is over the internet, the calls made to/from any Skype client or another Skip2Pbx are peer to peer and free of cost.

Implementation:

Skip2PBX is a software only solution that is installed on a dedicated server that has to have some kind of Internet access. The installation CD makes it easy to setup the application. If the PBX is not sip enabled then a Digium card with the kind and number of ports needed is added to the server to provide the interface to connect to the PBX. The kind of server, the Internet bandwidth and number or channels available in the PBX defines the capacity of solution.
Skip2PBX gateway should be connected on the LAN be behind a firewall to avoid being utilized as Skype relay node by other Skype users.
Please contact ABP Tech for more details on equipment needed, Skip2pbx demo license and for help configuring it. 

How does it work?
All VoIP traffic is routed thru a special SIP device connected to the local network that has the smarts to monitor the connection with main IP PBX, and when it deems the main IP PBX unresponsive due to network failure it will take over the call control intercepting the SIP messages and re routing them. Of course it is required to have local landlines (not Voip) in the remote office connected to the smart device’s FXO port(s). Calls between Ip phones in the same office would be routed within the LAN, emergency or any call to pstn would also be routed to the local landlines.
When the connection to the main IP PBX is restored the calls will start flowing thru it again as usual.

Implementation
As explained above, the smart SIP device is connected to the LAN and all IP phones configured to send all the traffic thru it. Two products fit the description for this smart device: one is the Intertex IX78 which is also a router and comes with one FXO port only, since all (voip and non-voip) traffic of the branch office flows thru it, the Intertex add the benefit of QOS and packet prioritization for VoIP calls; the other smart device is an Audiocodes Gateway configured for remote survivability that can handle 4, 8 or more local lines.
Please contact ABP tech to get the hardware and technical help needed.

How does it Work?
Most of new PBXs are already capable of handling SIP trunks, however in this case we are dealing with an existing PBX that is still working fine and the company does not want to replace it yet.
It won’t be required anything other than configuring the traditional Trunk on the PBX; the rest is all transparent for it. There, the call is converted to SIP and sent over the IP network to the SIP trunk provider which is in charge of terminating the call on the PSTN.
When an user wants to make an outbound call, it will be first handled by the legacy PBX that routes the call to the trunks connected to the Voip gateway also located at the company premises. This VoIP gateway will convert the analog or TDM signal to IP signal. This signal will flow over the IP network, usually the public Internet, to the SIP trunk provider’s network which can route and terminate calls directed to any PSTN number. Inbound calls follow the exact opposite direction.

Implementation
As shown in the diagram, a VoIP gateway is inserted between the Internet link and the PBX. Implementing this solution is fairly simple; however, the following have to be taken into consideration:
The number of trunk channels available in the PBX and the kind of bandwidth will define the supported number of concurrent calls that can be sent thru the SIP trunk. The Voip gateway can be analog or digital, again, depending on the kind of trunk ports available in the legacy PBX. If it is analog a FXS Voip gateway will be required.
The Internet link will carry the voice calls and rest of data from the company’s network therefore the required bandwidth needs to be carefully calculated and, in many cases, QOS/Voip prioritization policies need to be implemented in the company’s Internet router. Please contact ABP Tech for more details on equipment needed and help configuring it.

How does it work?
This applies to legacy PBXs. It won’t be required anything other than configuring extension numbers on the PBX; the rest is all transparent for it. The analog extension port of the legacy PBX is virtually extended with the use of an FXO VoIP gateway connected to it and an FXS VoIP gateway in the remote site. When the phone in the branch office dials anything, it would be like the phone is connected to the PBX and dialing. When the PBX rings the extension port in the PBX, the remote phone will ring.
The pair of gateways FXO-FXS converts the analog signaling and transports it over the IP network. Once the call is established the audio will be delivered the same way.

Implementation:
In order to implement this solution it has to be defined the number of remote extensions and based on that use gateways with the same number of ports. We also have to have the same number of extension ports available in the existing PBX. Additional channels might be needed if conference or consultative transfers are performed at the remote site. We assume that there is a reliable IP connection (over the internet or private) between offices with enough bandwidth to support the number of channels needed. Please contact ABP tech to get the hardware and technical help needed.

How does it work?
The fax machines do not change so we need to adapt the Voip network to support them.
Fax machines have to be connected to VoIP FXS gateways that support T.38. The gateway will recognize and decode the fax tones and instead of sending it as digitalized audio stream it will send the picture information as data packets. In the remote end there must be other T.38 enabled endpoint that will receive the data and reconstruct the audio tones before inserting to PSTN.
The T.38 protocol contemplates packet management techniques to enhance the quality transmission for the fax messages.

Implementation:
In order to implement this solution a VoIP fxs gateway with T.38 support is needed. The SIP trunk provider must support T.38 as well. If the IP-PBX does not support T.38 or at least act as T.38 pass thru then the VoIP call must flow directly between the endpoints that support T.38. Please contact ABP Tech for more details on equipment needed and help configuring it.

Why do we need IP PBX with Analog PSTN line?
In old times, the PBX was heavy, large and very expensive. It required a lot of hardware to perform the switching by closing copper circuits to connect calls. Nowadays, the IP PBXs are software based, the switching is logical and only IP data streams are peered when connecting calls.
However, there is still a need for interfaces to connect to the conventional local PSTN. Many companies will prefer to maintain their conventional trunks to not depend 100% on SIP trunks or the Internet link because consider it not reliable enough. In other cases, the only option to maintain business phone numbers is thru conventional trunks. Anyhow, it is not a bad idea keeping at least one or two landlines for survivability and emergency calls.

How does it work?
In the diagram shown above the Extension 1 to N are the IP phones which are connected to the LAN. These phones are registered with IP PBX server so it knows what IP (SIP contact or AOR) would direct the calls to when the extension is called.
The calls from the external PSTN first arrive at the VoIP gateway. There the call is converted to VoIP and sent over the IP network to the server which ultimately directs the call to the desired extension. Outbound calls follow the opposite path.

Implementation
Many appliance IP PBXs already come with such interfaces (FXO or T1/E1 ports) but software only based products will require an external VoIP gateway that would interface with conventional trunk lines and the IP PBX. Such gateway is configured to forward incoming pstn calls to the IP PBX’s IP address. For the IP PBX, outbound call rules or dialplan would route calls to the gateway’s IP, just the way a SIP trunk is configured.
Sometimes it is a bit of a challenge to get some not so conventional pstn T1/E1 working fine with the E1/T1 port in the VoIP gateway. Sometimes analog trunks will require fine tuning in the analog VoIP gateway to get it to detect disconnection signals, to eliminate echo or to get the right audio level. Please contact ABP Tech for more details on equipment needed and for help configuring it.

 

1. Don’t serve up alphabet soup- Real customer communication is more than acronyms 

Telephony technology is full of acronyms. You may know what you mean when you talk about DDoS (Distributed Denial of Service) and DECT (Digital European Cordless Telecommunications) but your customer, the SMB owner, considering this major purchase may be overwhelmed and confused. Worse still, they may be too proud to admit this lack of knowledge. As a result you may miss an opportunity to discover what the client really needs from their new system and so what extra things you could sell them.

In a world of IT and telecoms professionals where there is a short acronym for everything and very few explanations, you will stand out as a true expert if you say clearly what you really mean.

2. One size does not fit all - Offer products based on your client’s needs 

There is no problem without a solution. It is your job to listen carefully to your client as they describe their business and its practices so you can determine what they actually need instead of what you assume you can sell them. And don’t offer them a product based on their current broad band. Make sure they up-grade their Internet connection to match the job to be done, and not vice versa.

For instance, a growing company that has various telephony-connected devices that operate in analog mode (e.g. a building security alarm, credit card machines, fax machines) will need an IP PBX solution that can accommodate fax lines and their “regular” plain old phones.

Other companies want to make a large number of concurrent calls, have sophisticated call routing and voice prioritization features, and a system that links their tradition PBX traffic to the Internet. They will benefit from a gateway product that offers greater capacity, gives good sound quality and can be integrated with other applications.

3. Offer a complete solution - Sell more than just features and savings on long distance call 

Make your sales pitch have real meaning. Tell your client what a feature really does and why it’s important for them to have it to accomplish their business goals. For instance:

Quadro feature: “There are more FXO ports.” 

Customer benefit: “The FXO (regular analog telephone) ports make it possible for you to continue using your existing analog phones, fax machines and other analog devices. This way you can integrate the features of a new, IP telephone system with equipment you already have.”

Quadro feature: “You can buy keys in 16-port increments.” 

Customer benefit: “Buying keys to increase the number of available ports is an inexpensive way to add more capacity (= internal telephone lines) to your system. Your phone system can grow with your business, without having to buy additional hardware.”

Quadro feature: “There is a built-in capacity for failover calling.” 

Customer benefit: “We can configure your system to default to your traditional land line in case your Internet connection fails. This means that people in your office will still be able to make some calls until the connection is restored.”

4. Always be prepared - A good installation plan will make the configuration of the system much easier for you and your client. 

Choose your options; a quiet weekend on the lake or golf course or 48 hours in your client’s offices trying to discover what went wrong with their deployment. Good planning makes the difference here, although many people in the industry say that good pre-installation planning is unusual.

First and foremost: get trained on the product you’re selling the customer. No-one wants to see their ‘IT professional’ take the equipment out of the box and then act like he’s never handled one before. Knowledge is indeed power - get some.

In the meantime, here’s a handy checklist of other key elements that make a successful deployment: 

Engineer the LAN to limit latency to below 200ms.

Have a protocol that recognizes and prioritizes voice packets.

Use full duplex, non-blocking switches.

Install business grade routers, cable & related hardware.

Use voice-capable modems without firewall.

Deploy quality IP or analog phones.

Arrange for business class DSL (or upgrade to ISDN or E1/T1, if possible).

Ensure that the Internet service is connected to a private backbone.

Install quality IP PBX hardware.

5. Be safe and secure - More security now means fewer regrets later 

In a recent study among a group of VoIP users in the US, 40% of the respondents said they did not have specific plans to secure their VoIP deployments. 

The endpoints of your client’s VoIP system are vulnerable to attack. Among those areas that are now vulnerable to unauthorized access, viruses and worms are: operating systems, Internet protocols, applications and management interfaces of VoIP phones, desk-top computers and laptops running softphones.

It sounds scary however, it also means that there is a lucrative market for those who sell VoIP security products and services to their customers. Network security has to be part of a successful deployment. You can be a hero to your client by offering them the protection they need.

6. Plug and Play does not mean Install and Leave - Be there for the change over and train the end-user 

There are few things more traumatizing for employees than the changeover to a new system. Add to that the fact that VoIP phones may require special attention to function in a manner that is familiar to the new user. You may have installed a great solution for your client’s communication needs and you may have done your homework for the configuration, but don’t assume that the people who will be using the equipment will be able to understand how to use all the new features without help. (And don’t assume that anyone from their IT or facilities department will have trained them either.) Have your technician in the client’s office for the first day to make sure that the system is functioning and offer to train a couple of key people in the company who can then help others adjust to the new technology.

6 ½. If you sell only boxes, you are missing the chance to make much more money 

Congratulations. You’ve really listened to your customer’s communication needs and explained to him effectively the best solution and why it’s the best for his business. You’ve planned and executed a successful installment with a minimum of disruption and you’ve offered knowledge and reassurance to the new end-users who are using the new equipment for the first time on the Monday morning. You are a VoIP hero and your name will be a legend, but other than the thanks of a grateful customer and their great recommendations to their friends, what more is in it for you?

You cannot make good living just selling boxes; you can earn much more by selling packages that give you an ongoing future revenue stream while providing your happy client with continuing services and upgrades.

Money making scenario: 

Charge fees to evaluate and prepare your client’s network for the VoIP installation. Supply the voice hardware, Internet access and customization of features in the form of a monthly fee instead of a one-off charge. Training new users, and network acceleration and optimization can be written into the plan, as well as providing Voice security applications and services. 

Your client gets a complete communications support plan with personal attention that can prevent problems and you get a regular check every month – for years. Profits go to those who plan.

Warren Sonnen 

Epygi Technologies 

 2. In trunk definition select RFC3325 as shown below

Here is how the INVITE goes to that trunk, it shows the right caller ID in the From field:

INVITE sip:19728311600@192.168.0.243;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.7:5060;branch=z9hG4bK-5c6a45d05803349c98c47c597c6e3e4c;rport
From: “Henry Castillo” <sip:+19728311600@192.168.0.243>;tag=1049020745
To: <sip:19728311600@192.168.0.243;user=phone>
Call-ID: 0eba8451@pbx
CSeq: 15581 INVITE
Max-Forwards: 70
Contact: <sip:+19728311600@192.168.0.7:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: pbxnsip-PBX/2.1.10.2474
P-Asserted-Identity: “Henry Castillo” <sip:115@localhost>
Content-Type: application/sdp
Content-Length: 319

Henry

Technology and Business cycles are moving faster than ever. The world around us changes so quickly that it is key for all of us, Resellers, Vendors and ABP, to step out and re-sync to improve our business approach.  IP Sizzles 2008 will be a very interactive event in a workshop format that encourages attendee participation. Like our economy some areas are doing great while others aren’t. Some resellers are swamped with work and sales are booming, while others have flat sales. IP Sizzles lets you learn from the Best. What solutions are selling today? How are others growing their revenues? What do you need to change in order to be in sync with today’s market requirements?Twelve years or so back, when I was in charge of sales and marketing at another company in a different industry, I went to a conference and had an unexpected session on sales training. That session made an incredible impact on me, and in consequence, I was able to change the behavior of my entire sales team. The sales trainer I met then, was Dave Kurlan. Dave and I became friends, and his methods, and techniques had a long lasting effect on the sales of that company. We became friends and Dave had a long lasting effect on the sales of that company. I find that in many areas in technology, sales skills are still an afterthought. We invent the better mousetrap and think the customers will come. Most technology companies could get an unbelievable boost with just a bit of help in the sales skills area. For IP Sizzles this year, I invited Dave Kurlan to give us a half day sales presentation, and I am sure it will have the same sales boosting affect for you, as it had on me and my team in 1995. We also asked Dave to provide two follow-up Webinar sessions exclusively for resellers that attended IP Sizzles, so you can have your entire team get a firsthand experience of Dave’s practical and intuitive training. All in all, that will give you about 5 hours of sales training by one of the nations best specialist in the field.

We are committed to the reseller channel, and our partners’ success. Our partners believe that IP Sizzles is one of the most important contributions ABP can make to help grow a strong IP reseller community. You can only benefit from it, if you come. So I really want to encourage you to participate in this reseller meeting, that is much more than the sum of an exhibit, IP Technology presentations, and training. For ABP it’s the soul of channel distribution, and the basis of our relationships in the industry.

IP Sizzles 2008 will be two days packed with information sharing through case studies, interactive roundtables, panel discussions, and exciting keynotes. The exhibit hall will be packed with new products. There are many networking opportunities, half a day of top notch sales training, a full day of technical training, and more. IP Sizzles includes a lot of social networking time during breaks, meals, and after the sessions. This allows you to meet vendors, Industry Leaders, our entire team, and have a lot of dialogue with your peers.

We know you will find answers to technical issues, form new relationships, discover business opportunities and find the seeds to new ideas that will help propel your business to the next level, prepare and define your future.

Why do we do it?… We do it for you, and so should you!! SEE YOU AT IP SIZZLES ’08!!!

- Robert Messer