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Asterisk Servers

Main portion of the advantages of the Asterisk server appears from it being an open source downloadable product. Most of the users, especially business people, can find it difficult to understand the idea behind this principle, or free software, but a business models behind open source are different and specific, as in case with Asterisk server.

Asterisk IP PBX benefits

ITs source code availability provides assurance for business continuity

The question you would like to ask as owner of a business, considering on investing in any communication solution to your business issues, apart from costs, is if the organization behind the solution has a solid financial platform. Simply, if it is available in scheduled future to be sure that support is provided and communication solutions continue to be upgraded.

Once the question comes to a server to be utilized, there are 2 points. First of all, Digium, makes its revenues from trading servers compatible hardware and devices. While finding a market for their service or product, the attitude that Digium has used in open source solution, is very alike with the philosophy of many other companies. Simple examples are when something is offered or given away for free by an enterprise:

* Mobile carrier companies issuing discounted mobile phones on contract.

* Companies that issues free Primus cooking stoves, charges for the gas used in theses stoves.

Meaning that the Digium Company has a concrete business model behind Asterisk server, and it is almost unlikely that it will leave the scene in the future. Another aspect of the application is that presence of warranty, if Digium will abandon business, the open source code would still be available for download, or other entities, which are interested, are going to continue with the development and enhancement of Asterisk server concept. In modern world, where brand business tags have a short life cycle, there is absolutely no assurance that a proprietary supplier will stay in business for certain time.

Asterisk's server open source code and use of integral hardware means no proprietary lock-in

Additionally to the fact contract for Asterisk servers are available with no advanced payment, they have been created to operate with basic computer hardware giving you the chance to choose any hardware from any supplier. Even more, you can reuse the old hardware with the Asterisk server! Moreover, specific hardware you might require to obtain just to integrate it into your landline or cell phone systems, is based on common specifications, meaning that there is a plenty of compatible hardware vendors. And all that leads to the fact that the needed costs are low and you have excellent flexibility in your application.

IBM started the revolution of PC by releasing the open specifications for its first ever PCs, providing the entire industry and philosophy developing it around supporting and supplying PCs. If IBM had taken a different proprietary way, the computing world would definitely look totally different.

Asterisk server is freely and easily extensible

Next advantage benefit of using it, is that it is extremely easy to extend. It is extensible in hardware and software elements. From a hardware perspective, it is simple to extend an existing Asterisk server solution, without having to eliminate a hardware you have already installed on.  In case you look forward to connect several IP Phones and forward cheap phones calls via VoIP you can get a handy Asterisk GUI elaborated with supportive documentation and quick learning cycle. You can extend a PC with attached devices, or reuse components in a new PC, so too with your Asterisk server. You do not need to throw away the old PBX hardware in order to acquire innovations.

From a software perspective Asterisk server is really easy combinable into 3rd party programs. The core of the open source concept is the desire to freely combine with other programs, and, in our case, to allow other programs to be integrated with it.

Asterisk Servers asterisk servers

 Does the SQL Server 2005 Management Studio tool interpret the TSQL differently than the SQL Server 2000 Query Analyzer tool? The script contains a block comment like. Running this script in Query Analyzer connected to a SQL Server 2000 server generates an error. Running this script in Managment Studio connected to a SQL Server 2000 server does not. One and the other tools were connecting to the same SQL Server 2000 serve. Are all client tools doing some interpretation prior to sending the code to the server?

Then, in oracle there is a provision to add comments to a table, as we aware.

asterisk servers

Thank you for your suggestions and hints. Besides, thank you! Does the SQL Server 2005 Management Studio tool interpret the TSQL differently than the SQL Server 2000 Query Analyzer tool? The script contains a block comment like.

Running this script in Query Analyzer connected to a SQL Server 2000 server generates an error. Running this script in Managment Studio connected to a SQL Server 2000 server does not. Remember, all tools were connecting to the same SQL Server 2000 serve. Yes, that's right! Are all client tools doing some interpretation prior to sending the code to the server? Finally, in oracle there is a provision to add comments to a table, as we aware.

Hey can anybody help me in using loops to get some values from a database with stored procedure in SQL SERVER hi thanx a lot I have learned a lot more from ur web sit and ur youtube videos bout I have a short request what will be in the event I want the product id in the subsequent example equal max not a specific value plz replay me via mail Thank you for your suggestions and hints. Oftentimes thank you!

Weekend and thank goodness no exception to the rule, once a year I give my blessing to the wife to go away on a long weekend with the girls and mostly I try to call in small amount of childbaby minding favours from my parent & mama/inlaws and this is. Furthermore, last time I was given lately of peace I wrote a Trixbox/Exchange 2010 integration guide, the emphasis was on this becoming the 1-st in a series of howto's -however this in no circumstances actually came to fruition, the reason? Asterisk + friendly UI = rubbish nasty terrible…so from here on in I had chosen to move to AsteriskNOW.

asterisk servers

Trixbox is a good distribution of Asterisk, however it does break particular Asterisk standards and you can not beat an excellent ol' command outline -yes in Asterisk's case the command straight is easier compared to a web interface. It's a good idea to plain old enough Asterisk? Anyways, asteriskNOW makes light install work and I'm by no means a Linux guru! You can still decide on the FreePBX front end -but we will choose to not go down this grim path -trust me on this!

Sounds like a tall order right? I'm sure you heard about this. Bad. It is reasonably straight forward and I will endeavour to document the 'endtoend' setup process, with AsteriskNOW and Lync Server 2010. Skype as you may or may not be aware offers 2 SME level VoIP integrations.

asterisk serversasterisk servers

That's where it starts getting really interesting, right? In case you are always running a Asterisk based PBX you will possibly want to see the difference. From a big level it comes down to the subsequent. One last caveat until we get on with the good stuff.

Have you heard of something like this before? Download a copy of AsteriskNOW, I had opted for the 64bit version here, whilst this is downloading, we will setup the VM.

Step by step' Microsoft Lync Asterisk, skype as well as 2010 installation/integration guide

Have you heard about something like this before? Download a copy of AsteriskNOW, I got opted for the 64bit version here, whilst this is downloading, let us setup your VM.

You'll need to search for your settings and add one hardware component, the legacy network adaptor mentioned earlier -and be sure this is connected to your virtual network we need to start the VM, unto we kick off the install.

Select yes, to accept partitions creation and wiping of info The default partition scheme is fine, select next. Set your select next, create or location a root password then click next.

Known it is Asterisk v1. Then once again, freePBX v2.

That's right. Right now when I try to call externally thru my Tanguay phone I get a Call unsuccessful. I'm sure it sounds familiar.should not complete the call due to restrictions on your account error.

When I try to call thru the Lync Client I get this. All in all, things 1st is there a SIP trunk successfully established between Lync/Asterisk? Notice, in Asterisk you can check this via the CLI -on Lync the server event log will report trunk related errors. Just think for a minute. Ensure your PBXIAF is set to allow TCP -this is not enabled by default.

Make sure you write some comments about itbelow|in the comment section. There lots of ways this could be achieved, my lab setup was configured to automatically prefix Lync PSTN routes with 9” as per my config -this will route via SFA. Seriously. Calum MacRawe -did you ever get this figured out…I'm having the exact same problem on Lync final release.

This guide is absolutely awesome. Lync step by step guide to get an essential Lync server up and running and I'll apparently use this to get some awesome integration going. Chances are you had constraints with your call routing, specifically within Lync. Does your Lync enabled user have an extension assigned? When calling what actually is the error reported within the Asterisk CLI -type asterisk -r on your Asterisk command outline. That's right. Let me understand how you get on.

Used this article and have got skypeout working fine, however I had got an online number setup on the skype account but having some constraints routing the inbound to the lync client. Hi Craig, now when you are trying to get assistance on account of your flattery then you are going the right way about it!

Off have you given the extension number 2001 to your Lync client? Nevertheless, next how have you triggered your trunk, the busy tone mostly indicates a trunk/routing concern. From Lync I can call extensions on Asterisk and AudiocodesMP.

Yes, that's right! From AudiocodesMP I was not able to call any extensions on Asterisk until i've configured 'NewCSAnalogDevice' account in Lync for AudiocodesMP. Now using +900 as CallerID I can call any number from AudiocodesMP. When calling +900 from Lync I can reach AudiocodesMP. When I call on +900 from Asterisk.i've noticed what when I phone +900 from Asterisk, lync is trying to ring +900 extension on Asterisk after AudiocodesMP, while monitoring Asterisk SIP log.

asterisk servers

It is when use its TEL, cSAnalogDevice account for Asterisk. Of course callerID when I call from Asterisk to TEL. Seriously. This Call is routed. There's some more info about it on this this case I'm loosing my original Asterisk CallerID and this is not acceptable. That's to +900 still fails. A well-known matter of fact that is. How is it possible to solve my issues. Asterisk to AudiocodesMP through Lync and keep original CallerID?

asterisk servers

Nonetheless, wow Igor, this one threw me a little For you, why are you not using the AudioCodes as a gateway for Lync and Asterisk? When you took the technique you could establish cross extension capability via a Lync to Asterisk trunk.

Loads of info can be found online. Asterisk and Lync to share the same extension? You should take this seriously. What I'd like to accomplish is to ring your existing Asterisk IP desk phone and ring/show the calls on the users Lync client, basically a simultaneous ring using the same extension. Did you hear of something like that before? can not figure out methods to do it or when it's even manageable. Would a simultaneous ring to yor Lync extension not achieve this?

You can find more info about this stuff here. Does Skype Connect or SFA allow 2 way audio communication between cloud of Lync users and cloud of Skype users? Then once again, what about the same question about messages exchange between that clouds?

Lync is unable to normalise the calls for 'Lync2Skype' purposes, skype does not allocate cloud based DIDs. Potentially you would be able to setup a speedial within Asterisk and assign a number that may be reached via Lync but this starts to get complex. The biggest win is able to utilse Skype's PSTN for outbound and for inbound you can use Skypein or your Skype ID for Skype based clients.

Cost optimisation via Skype is not a question, surely, we got a comfortable long distance prices from the provider. This requires Skype client installation and interfere with security policies, poser is that users are requiring Skype to chat, use or talk videoconference with contragents and to minimise 'longdistance' expenses. We're stuck with Skype interoperability, lync 2010 thereafter. Meets all the internal corporate needs.

Potentially, there're number of solutions to integrate Skype and Lync, skystone as well as for sake of example Skystone Video. Virtually, will evaluate it. Stating the obvious 1-st, have you perfectly assigned the extension no? However, next have you looked at the Asterisk logging? Type asterisk -r via the command prompt. Besides, is the trunk up?

Ok, and now one of the most important parts. Hi, thanks for a big guide, just walked thru this and got it all working, thanks! Essentially, lync and mobile -> Lync where fine.a good poser I ran to was that I got no audio from the Skype client to Lync, X lite ->. The fix I searched for was to add the next to chan_skype.

asterisk servers

Of course, unable to connect SIP socket to 192. Connection refused Thanks for posting your workaround Stephen -was this not enabled under the patronage of default?

You should take this seriously. Apologies for not getting back to you sooner and thanks for your previous message, given that Windows Firewall is disabled you may want to check the ports -often there could be some confusion on incoming/listening ports. Now please pay attention. TechNet documentation for default port definitions.

Keep reading! Skype -okay so it was a workaround and PIC should be far I do not have a Chan_skype. You should take it into account. Is this something I will create manually?

This is always created when you add SFA -is this installed properly? Far as I can tell. License Key for the SFA and that didn't give me any errors and it created the License File.

You see, this would definately be created as an install fraction.

GREAT POST! Thanks to your rough work I got my environment up and running. That said, lync users can call 'X Lite'. XLite users can call Lync. Lite users can connect out via SIP. With that said, lync users can connect out via SIP. All working like a charm. You should take it into account. Not that it happened readily. Just think for a fraction of second. Some rethinking were required. Reloading the configs is much way faster. Specifically when testing unusual dial plans.

With all that said. Thanks Alex, appreciate you sharing this info for various different readers. It is lync Asterisk' element, I solely have my doubts in the subsequent.

With that said, when somebody has any suggestion on this to be of consideration I will be big full in the event you share until I go for deploying.

Did you hear about something like that before? R2 SP1 Hyperv server. Have you heard of something like this before? The Linux Integration Tools don't install really! Centos two fine but when I get to the integration tools I will not access the command outline nor install the tools. Generally, any representations?

Lots of information can be found online. Hello Adam pls I will like in case u can assume to me a solution to this I got this from eventvwr The Mediation Server service has got a call that does not support comfort noise. This event is throttled after five calls from a single Gateway peer.

Now regarding the aforementioned matter of fact. The comfort noise constraint is unlikely to be the root cause, suppose as a 1-st port of call you look to the Asterisk routing error. It is access the Asterisk command threshold and examine the logging generated wen you try to initiate a call failure. Now pay attention please. The command straight line could be accessed under the patronage of typing asterisk -r via the terminal.

Within the Asterisk CLI type sip show peers -is your Lync trunk up? Try the Asterisk CLI, this is enabled with the help of typing asterisk -r from your Linux terminal.

I'm sure you heard about this. HI Adam I am so sorry for disturbing you.

While something is stopping your Lync trunk from being established -this is the root cause, you config looks good. Ok, and now one of the most important parts. Not sure in the event I got asked, do you had Windows Firewall enabled, could this be blocking the port? Is 192. Notice that iP address of your Lync server? Will I merely revert and try to use port 5068 and see what happens?

Generaly, i should definately assume you revert to 5068, I got this configuration setup and working with the default port configuration. Needless to say, fantastic article. Now I can eventually get some real handson experience with Enterprise Voice.

Adam, attempted to purchase license for AsteriskNOW and it seems that it is no longer accessible? It's a well am I missing something on there webpage? Error. SIP/'Lync_Trunk 00000004' is circuitbusy Please help me….

Lync trunk, at a guess you had some sort of connectivity/performance related concern impacting the Asterisk ->. It's a well the client gets a faster busy. That said, amidst the errors I see in event viewer on the server is There was no response from a gateway to a OPTIONS request sent by the Mediataion Server. Any suggestions?

You see, thanks for the feedback Mike. Anyways, since not looking to it is your trunk UP, xecs I see it is famous? Thanks for reply Now Asteris is Workinf Fine and I had configure Elastix GUI for LYNC.

Does everyone have a clue where to begin troubleshooting this? Cause I'm clueless. In reality, the best stuff that differs from your Lync conf is the regexp for public calls which is ^ later. Often, swedish mobile phone number is +46734123456.

Is it possible to please let me understand SIP vendors which I can use to call PSTN. Please let me understand the configuration. In case I understand this configuration carefully you got 2 Trixbox trunks defined -this is not neccessary and will confuse things. Lync accordingly.

Stepbystep Microsoft Lync Asterisk, 2010 as well as Skype installation/integration guide | I'm a UC Blog… I apologize for apparently a dumb question here. Of course sIP/IAX/Lync, is and PSTN Now pure presence/IM platform; Asterisk based solution is about telephony. What help do I get with integrating the 2? Cause I can have OpenFire server on my Asterisk box providing IM/presence for these same extension via AsteriskIM plugin, when the talk on is phone presence and IM's then I still don't understand. Please support me in case I am incorrect or missing something. IF there is a feature I can further refine my solution, I am open to extra info.

Bria which has integration with Outlook, asterisk, contains presence, click to dial or even IM features What's having purpose a massive product like Lync integrated with Asterisk? From ease of it, deployment, complexity of management as well as licensing cost seems to be an overkill. Lync is that lot more than just presence and IM a/v conferencing and collaboration, since ffice Communications Server 2007 the platform has mostly incorporated voice. Mostly, oCS/Lync integration with Asterisk can deliver plenty of aids, such as existing leverage Asterisk PSTN breakout.

Perhaps one of a few Lync Asterisk integration guides out there. Totally newest to Asterisk, lync since past couple of years. In any case, alex did in Post 73 below e set this up with a Asterisk Sip Trunk Provider which unfortunately doesn't work with Lync. Ok, and now one of the most important parts. Is the trunk up betwixt your Asterisk server and Lync? I'm sure you heard about this. Sip show peers will confirm, as you previously stated.

Thank You for your reply. Yes it is up as reported by the sip show peers command on TCP when dialing from Lync to 'X Lite' I get 480 temporarily unavailable on Lync Mediation and Sip 404 Not figured out when I call from 'XLite' to Lync Try asterisk -r at the command straight line and replicate a call to see what's going on -it might be a route constraint?

PIAF purple and trying to move off my old enough pbx that I had working with Lync merely fine.

I'm sure you heard about this. In PIAF purple sip. Lync is not talking to Asterisk, from this it looks at though your Asterisk server is talking to Lync. Check your Lync SIP trunk out within topology builder -is it set to 5060?

On top of that, when so whats the average capacity per sip trunk to help multiple call volume.

Just think for a minute. Say I want Lync users to utilize there client as there soft phone. That user wants to forward calls from Lync to the cell phone. That call will route out my Asterisk box. Asterisk extension and have Lync set to Forward the call. It translates the call back to Asterisk as a famous extension from an outside source in this case that is attempting to say hey Im one of your extensions please forward this call.

Pretty good work around I searched with success for wich is not ideal but does work is when you don't set a password on the extension. This way Asterisk does not try to validate a reputed extension from outside and help it to pass the traffic. Anybody try this or have an exclusive method of engineering it to work with the Asterisk extension and having success forwarding Lync calls out?

Absolutely, you can have a range of DDIs assigned to a trunk. Whilst, capacity is based upon network capability and any associated service provider restrictions. Have you heard of something like this before? Last at last, okay so I have got everything back on 5060 and now all seems to be lucky once more.

Surely, my last nudge here's. It is pBX to the Lync Extension.

Furthermore, sIP to dial my cell phone number over the ExternalProvider trunk will like to give a large thanks to Adam Jacobs who wrote ways to integrate Lync 2010, asterisk and as well Skype as inspiration for this post. Now please pay attention. This entry was posted in How To, lync by Jaredg. You should take it into account. Bookmark the I'm trying something identical with PIAF. PBX with a Google trunk. Now I'm trying to get the SIP trunk between Lync and the PBX up but simply having a horrible time! Lync? Cause for me life I can not get the trunk to come up.

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