Planet Asterisk

Hi. We have some newer i2002 phones that do not seem to work with the unistim driver. Model is NTDU91 and firmware is 604DAD. Questions 1. What version of hardware / firmware was the unistim driver written for? 2. Is there a way to download my firmware and compare the differences and update the code? 3. Has anyone had an i2002 phone working with the unistim driver? Thanks Kevin
kesau () at 2008-07-24 18:18 GMT

July 24, 2008 05:30 PM

Apparently, in Bangladesh if you try and resell VoIP services out of your house, you're apt to get raided. Hope what happened to American Eliadah "Lia" McCord (forced to smuggle drugs, caught by Bangladesh airport security, gets 30-year sentence, but eventually pardoned thanks to help from Gov. Bill Richardson) doesn't happen to people who run VoIP services out of their house.

I happened to see the Eliadah "Lia" McCord story a few nights ago on TV. It was a tragic story that ended up well, but not before spending 4 years in prison. In Bangladesh drug smugglers are typically executed (hung) and they don't have the lengthy appeals (death row) process that happens in the U.S. and other countries.

Imagine getting executed for operating VoIP. Though I doubt even Bangladesh is that extreme. Probably just a 10 year sentence.

check it out:

Tk 1cr VoIP equipment seized in city

Voice over Internet Protocol (VoIP) equipment worth around Tk 1 crore were seized from two places in the city early yesterday.

A committee formed to inspect and detect telecommunication installations conducted separate raids on two houses at Kanthalbagan and Paribagh from 12:30am to 4:00am and seized the equipment.

Officials of Bangladesh Telecommunications Regulatory Commission (BTRC) and Rapid Action Battalion (Rab) are the members of the committee.

The equipment include 120 tellular of various models, 7 gateways and one 32 port channel bank, says a BTRC press release.

Tags: Bangladesh, Bangladesh Telecommunications Regulatory Commission, BTRC, Eliadah McCord, voip

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Copyright VoIP & Gadgets Blog

July 24, 2008 02:30 PM

Vonage today announced two new low cost call plans for their Vonage UK subsidiary. According to Vonage UK, "Following customer research and reacting directly to consumer concern about increasing household costs and spiraling business overheads, Vonage has created two new fixed rate call plans."

They added, "Vonage subscribers report enormous savings on their monthly bills and comment on the speed and ease of swapping providers as well as installing Vonage. The new call plans fly in the face of increasing utility prices and the new £6.99 plan has been designed for the high percentage of Vonage consumers requesting more cost efficient plans for North America."

Vonage's £7.99, £14.99 and £18.99 plans incorporating up to 45 countries remain unchanged. The two new call plans offer Vonage's lowest ever rates and are called V-Plan UK and V-Plan US.

• £5.99 per month - unlimited calls to the UK (V-Plan UK)
- Premium features such as call waiting, caller ID, call diversion, voicemail, three way calling (normally billed as extras with other providers), are included as standard.

• £6.99 per month - unlimited calls to the UK, United States and Canada (V-Plan US)
- As above plus, for only £1 extra per month, unlimited calls to the US and Canada to include calls to US and Canadian mobile phones.

Here's a screenshot of the various V-Plan calling options. Click image to see the plans:


Vonage to Vonage calls are free. Also, there are no hidden costs with Vonage - prices are always quoted including VAT.

Vincent Potier, Managing Director of Vonage, said; "We recognise how important it is for customers to keep costs low and as predictable as possible - especially in light of the current economy. Our new plans enable customers to make worry free calls for as long as they want and show our long term commitment to listening and responding to our customers as well as offering the highest level of customer service and value for money".

Tags: calling plans, United Kingdom, V-Plan, VoIP, Vonage UK

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• Comcast - We won't block Vonage or any other VoIP provider - Jul 10, 2008

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Copyright VoIP & Gadgets Blog

July 24, 2008 02:30 PM

I missed the 8x8/Packet8 news on July 16th about its new hosted small office "key" system and plug-and-play IP phones. Joan Citelli, Director of Corporate Communications emailed me asking for a briefing, but apparently I never replied since her email was still marked as unread and nothing in my Sent Items. Email overload I guess. The news was about 8x8, working with handset maker Aastra Telecom to provide a key system to the SMB market,which is part of the new Packet8 675xi series.

I happened to come across Carolyn Schuk's article while surfing the web and came across her post about some 8x8 news that I missed. She writes, "8x8 is VoIP's Rodney Dangerfield. It just gets no respect." She has an excellent point and one which I wholeheartedly agree with.

Carolyn then lays out her case: "Consider how it stacks up against its far better-known pure-play VoIP competitor, Vonage: In the last five years, 8x8 revenues grew 460.3 percent while Vonage's grew 0.0 percent. 8x8 made $700,000 during the first quarter of this year. Vonage lost $8.9 million and is shopping for a $215 million refinancing deal to stay out of bankruptcy. 8x8 holds 73 patents. Vonage just got its first. Despite this, Vonage's stock price is $1.58 while 8x8's is $1.03."

She goes on to explain that the news coverage of the new Packet8 675xi series was sorely lacking, which sparked the Rodney Dangerfield comment. I'm guilty as charged, since I didn't cover the news. Though it wasn't for a lack of respect that I didn't cover the Packet8 news. Sometimes it's just impossible to cover all the daily VoIP news in addition to my testing of VoIP products, managing the MIS department as CTO, etc.

Well, better late than never. Today, I thought I'd give an overview of the new Packet8 675xi series, which is actually part of their Packet8 Hosted Key System Services. Perhaps most importantly, this offering supports "call appearances" commonly referred to as "shared line appearances" or SLA, which enables you to know when someone is using a line. It's a popular feature of key systems and one which is often difficult to reproduce on VoIP systems. Supporting SLA is often a key selling advantage when targeting the SMB which is used to call appearance functionality.

First off, the Packet8 675xi IP phone series consists of three models -- the 6753i entry level phone, 6755i intermediate phone and 6757i CT advanced phone. Essentially these are OEM'ed versions of the Aastra 53i, 55i, and 57i CT but with a special firmware load. Each model offers full duplex speakerphone functionality, programmable softkey appearances, LCD display screens, embedded XML browsers and up to nine call appearance lines. All models support Power over Ethernet and come equipped with dual auto-sensing switched Ethernet ports.

Here's pictures of all 3 models:




The Packet8 675xi series include intercom paging and direct dial from a searchable corporate directory. Prices for the Packet8 675xi series range from $129.99 for the 6753i to $349.99 for the high end 6757i CT model which includes a DECT cordless phone as part of the bundled offer. The 6757i CT model's built in DECT antenna allows the user to roam up to a 300 foot radius from the 6757i CT base telephone. The Packet8 675xi IP phones also feature corporate directory display and lookup, intercom paging, and shared line appearance.

The Packet8 675xi series of IP phones incorporates 8x8's advanced NAT traversal technologies. This allows users to simply plug the phone into any Internet connection and immediately make or receive calls without performing any network or firewall configuration.

The high-end Packet8 6757i CT includes an integrated cordless handset with coverage up to 300,000 sq ft. It has a large 144 x 128 pixel graphical backlit LCD display and 6 dynamic context-sensitive softkeys, and with its large screen it can take full advantage of XML based programs.

Lastly, and perhaps most importantly, the Packet8 675xi series use SIP trunking to Packet8's network infrastructure. All of the telephony functions such as transfer, conferencing, voicemail, etc. reside on the Packet8 network. Thus, you don't need any costly IP-PBX hardware at the customer premise - you just need IP phones. This can be a huge cost savings for SMBs looking for an inexpensive VoIP solution, especially as the costs and margins for IP-PBXs continue to shrink with growing price pressure from more competition and open source solutions like Digium's Asterisk.

Packet8/8x8 certainly has earned my admiration with some great products and services, a cool videophone, and more VoIP patents than you can shake a stick at! My 'respect' has been duly given.

Tags: 8x8, Aastra, Asterisk, Carolyn Schuk, Digium, IP phones, Packet8, Packet8 6757i, Packet8 6757i CT, Packet8 675xi, smb

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July 24, 2008 02:30 PM

After using the patch from Paul, I had some trouble with the cli - command. If i forgot the arguments (like: fundevstate change) asterisk crashed immediately. Second issue i had was that i always got a warning in the CLI. Changed both in the file and edited the comments. Hope this patch will help some of you guys. New download here: http://www.hdpnet.de/downloads/asterisk/devstate/func_devstate.c
dominic () at 2008-07-24 10:46 GMT

July 24, 2008 10:46 AM

salut les camarades: svp j'ai téléchargé la verison 2.0.1 d'asterisk-stat.tar.gz mais j'ai pas aucune idéé sur sa configuratio pour la faire marché correctement est ce que j'ai besoin de créer une base de donnée mysql? svp j'ai besoin de votre aide
mettichi () at 2008-07-24 06:55 GMT

July 24, 2008 06:55 AM

salut : j'ai besoin de votre aide j'arrive pas à connecter deux serveurs asterisk. le premier serveur à l'@IP 192.168.1.203 les extensions vont de 100 à 199. le deuxieme serveur à l'@IP 192.168.1.67 les extensions vont de 200 à 299. malgré que ma configuration est correcte je rencontre j'arrive pa à appeler depuis un serveur vers l'autre. voici ma configuration: 1er serveur à l'@IP: 192.168.1.67 iAX.conf [server1] type=friend user=server1 secret=server1 host=dynamic ; Nous obtenons l'adresse IP lorsque l'autre PBX s'enregistre context=incoming_training_centre_calls auth=md5 ; Securiser l'authentification disallow=all allow=g729 trunk=yes qualify=yes ; Nous activons le trunking extensions.conf [outgoing_training_centre_calls] exten => _1XX ,1,Dial(IAX2/server2:server2@server1/${EXTEN:2}) exten => _1XX ,2,Congestion ; En cas d'echec une tonalite de congestion est utilisee [incoming_training_centre_calls] exten => _2XX ,1,Dial(Zap/1) ; Appels provenant du centre de formation ; diriges vers le telephone du telecentre deuxieme serveur à l'@IP: 192.168.1.203* IAX.conf [server2] type=friend user=server2 secret=server2 host=dynamic ; Nous obtenons l'adresse IP lorsque l'autre PBX s'enregistre context=incoming_telecentres_calls auth=md5 ; Securiser l'authentification disallow=all allow=g729 trunk=yes qualify=yes extensions.conf [outgoing_telecentres_calls] exten => _1XX.,1,Dial(IAX2/server1:server1pass@server2/${EXTEN:2}) exten => _1XX,2, Congestion [incoming_telecentres_calls] exten => _2XX.,1,Dial(SIP/202) voici le message d'erreur quand j'appelle: Executing [102@internal:1] Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/2") in new stack [Jul 23 14:17:56] WARNING[4790]: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16384 -- Hungup 'IAX2/server1-16384' [Jul 23 14:17:56] WARNING[4790]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [102@internal:2] Congestion("SIP/vente-b761ee68", "") in new stack == Spawn extension (internal, 102, 2) exited non-zero on 'SIP/vente-b761ee68' -- Executing [200@internal:1] Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/0") in new stack [Jul 23 14:18:01] WARNING[4791]: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16385 -- Hungup 'IAX2/server1-16385' [Jul 23 14:18:01] WARNING[4791]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [200@internal:2] Congestion("SIP/vente-b761ee68", "") in new stack == Spawn extension (internal, 200, 2) exited non-zero on 'SIP/vente-b761ee68' -- Executing [202@internal:1] Dial("SIP/vente-b761ee68", "SIP/vente|20") in new stack -- Called vente -- SIP/vente-0a1752b8 is ringing == Spawn extension (internal, 202, 1) exited non-zero on 'SIP/vente-b761ee68' -- Executing [203@internal:1] Dial("SIP/vente-b761ee68", "SIP/commercial|20") in new stack [Jul 23 14:18:15] WARNING[4793]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [203@internal:2] VoiceMail("SIP/vente-b761ee68", "3000@default") in new stack -- vente-b761ee68> Playing 'vm-intro' (language 'fr') == Spawn extension (internal, 203, 2) exited non-zero on 'SIPvente-b761ee68' -- Executing [101@internal:1] Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/1") in new stack [Jul 23 14:18:21] WARNING[4794]: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16386 -- Hungup 'IAX2/server1-16386' [Jul 23 14:18:21] WARNING[4794]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [101@internal:2] Congestion("SIP/vente-b761ee68", "") in new stack == Spawn extension (internal, 101, 2) exited non-zero on 'SIP/vente-b761ee68' -- Executing [103@internal:1] Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/3") in new stack [Jul 23 14:18:25] WARNING[4796]: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16387 -- Hungup 'IAX2/server1-16387' [Jul 23 14:18:25] WARNING[4796]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [103@internal:2] Congestion("SIP/vente-b761ee68", "") in new stack == Spawn extension (internal, 103, 2) exited non-zero on 'SIP/vente-b761ee68' -- Executing [104@internal:1] Dial("SIP/vente-b761ee68", "IAX2/server2:server2@server1/4") in new stack [Jul 23 14:18:30] WARNING[4797]: chan_iax2.c:8815 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/server1-16388 -- Hungup 'IAX2/server1-16388' [Jul 23 14:18:30] WARNING[4797]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [104@internal:2] Congestion("SIP/vente-b761ee68", "") in new stack == Spawn extension (internal, 104, 2) exited non-zero on 'SIP/vente-b761ee68' [Jul 23 14:26:17] NOTICE[4300]: chan_iax2.c:8645 __iax2_poke_noanswer: Peer 'server1' is now UNREACHABLE! Time: 1 [Jul 23 14:33:08] WARNING[4305]: chan_zap.c:6685 handle_init_event: Detected alarm on channel 4: No Alarm please help me!!!
mettichi () at 2008-07-24 06:52 GMT

July 24, 2008 06:52 AM

July 23, 2008 11:29 PM

Check out ((Asterisk embedded systems))
admin () at 2008-07-23 23:05 GMT

July 23, 2008 11:05 PM

I have configured asterisk as registrar to test my code. Now I can register and make a call. I would like to add an additional header, say "P-Associated-URI" in 200 OK response from asterisk. Can we add an header in 200 OK response from ASTERISK ? If so, where and How ? Thanks Srinivas
srinivas_gowda (Srinivas) at 2008-07-23 19:26 GMT

July 23, 2008 07:26 PM

I need help setting up a separate tftp server with the files for Cisco 7960g phones to work on a asterisk based Switchvox system. $300 reward if you can get me running today, July 23 2008. neilyoungberg@gmail.com
neily () at 2008-07-23 15:36 GMT

July 23, 2008 03:36 PM

Each attendee will be responsible for their own travel, meals and lodging costs; the conference sessions will only have a beverage bar and light snacks. There will be free wireless Internet access in the meeting room and in the guest rooms at the Renaissance.

This year we plan to focus our efforts on media stream handling and codec (format) negotiations; at the previous two DevCons we have talked about these topics but not made any significant progress, and it's time to get the work done to improve Asterisk so it can do a better job handling complex media streams and changing codec requirements.

If you are interested in attending, send an email application to Kevin Fleming at kpfleming@digium.com including your name, your involvement with Asterisk (or related projects), and who is sponsoring your attendance (if any company or person is doing so). We will accept applications until August 15th, and then make the decisions about who we can accept based on their level of contribution and the space available at the event.

You can find accommodation and travel information on the AstriCon website at AstriCon.net

July 22, 2008 06:24 PM

email me at admin@ebiz-pro.com
priyanke (DaP) at 2008-07-22 17:13 GMT

July 22, 2008 05:13 PM

Looking to remove my linux server, so I need a very small appliance that I can run my Asterisk on. It only has to support 3 phones and small voice mail. I'm looking to lower my carbon footprint on what machine I run Asterisk on.
awysocki () at 2008-07-22 16:48 GMT

July 22, 2008 04:48 PM

Today we solicit your feedback on the new Software Update Service proposal for PBX in a Flash...

July 22, 2008 02:00 PM

Voip ATA can support Asterisk Introduction: HanLong Unicorn Analog Telephone Adapters/Gateway series offers affordable VoIP access devices based on SIP protocol. The Unicorn series offers the user rich functionalities and compatibility with most service providers. Company: Hanlong Technology Co.,Ltd Product: ATA Brand Name: Unicorn Certification: FCC/CE Service: OEM & ODM ATA Series: ATA:Unicorn 2112(2RJ45+1FXS+1FXO) ATA:Unicorn 2002(2RJ45+2FXS) ATA:Unicorn 2101(2RJ45+1FXS+1Lifeline) Advantages: A:High performance-to-price ratio, Such as FOB Price for Unicorn 2112 is USD43/Unit(Quantity :100-499) B:Easy to manage and scalable,support web config,upgrade via HTTP or TFTP. C:Our products can be modified according your requirements D:Auto-restart by watchdog E:Chance to get a free IP PBX(See the following promotion) Promotion: If you buy more than 50 lines of our voip products, we will present you one piece of IP PBX. All our voip products support Asterisk,especially FXO Products! Contact information: Contact person: Miss Grace MSN: grace_syy2008@hotmail.com Skype:grace_syy Email:grace@mail.hanlongtek.com Website:http://www.hanlongtek.com
hanlongtekvoip (Grace Shi) at 2008-07-22 07:12 GMT

July 22, 2008 07:12 AM

Voip Gateway can support Asterisk Introduction: The Unicorn Gateway support popular voice codecs and is designed for full SIP compatibility and interoperability with 3rd party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology,integrate a traditional phone system into a VoIP network, and efficiently manage communication costs. Company: Hanlong Technology Co.,Ltd Product: Gateway Brand Name: Unicorn Certification: FCC/CE Service: OEM & ODM Gateway Series: Unicorn 6004 (4FXS+ 1 Life line) Unicorn 6008 (8FXS+ 1 Life line) Unicorn 6040 (4FXO) Unicorn 6080 (8FXO) Advantages: A:High performance-to-price ratio, Such as FOB Price for Unicorn 6004 is USD59/Unit(Quantity :32-99) B:Easy to manage and scalable,support web config,upgrade via HTTP or TFTP. C:Our gateway can be modified according your requirements D:Auto-restart by watchdog E:Chance to get a free IP PBX(See the following promotion) Promotion If you buy more than 50 lines of our voip products we will present you one piece of IP PBX. All our voip products support Asterisk,especially FXO products! Contact information: Contact person: Miss Grace MSN: grace_syy2008@hotmail.com Email:grace@mail.hanlongtek.com Website:http://www.hanlongtek.com
hanlongtekvoip (Grace Shi) at 2008-07-22 07:11 GMT

July 22, 2008 07:11 AM

Paul Adams contacted me recently since he's been reading my posts about Asterisk and 'Microsoft OCS 2007' integration.

Paul wrote an interesting application that enables call queuing that 'respects' OCS presence. That is, if the agent's OCS status is "Busy", "Away" or "Do not Disturb", don't send them a call from the call queue. Leveraging Asterisk.NET & the Microsoft OCS development tools - in C# (Visual Studio 2005) - he was successfully able to control calls in Asterisk 1.4 based on any user's presence in OCS.

He wrote a simple test app that register's with OCS 2007 for a single or multiple users presence. Whenever a users presence changes - OCS 2007 informs his app directly.

Using an agi entry in the Asterisk dial plan, Asterisk asks his app what to do next. Then - based on the user's presence - the app tells Asterisk to queue the call or pass it to the user.

Eventually, he intends that the user can 'register' themselves for call queues (via a web page perhaps) - and this information will be used by the app to determine what calls should go to what users & if they are available right now or now. He explained that he intends to turn this app into a service and run it on his OCS or OCS Mediation server - and control incoming calls for his call center.

He told me, "I'm impressed with Asterisk.NET - and with the tools from Microsoft - they have made it really easy to monitor presence. It's more difficult to CHANGE presence - but I'm not so worried about that right now."

With information Paul sent me I was able to write a tutorial on controlling Asterisk based on an OCS user's presence. Credit goes to Paul for this tutorial.

Controlling Asterisk based on an OCS 2007 User's Presence Tutorial

This tutorial, although very basic, demonstrates is that it's easy to add 'OCS presence' awareness to desktop apps written in Visual Studio. Then you can control Asterisk using Asterisk.NET.

This is not intended to be a professional, server-based solution (because it uses the desktop Office Communicator client) - there is other MS development API's for OCS server interaction. Still, this provides some powerful presence integration with the popular Asterisk and OCS 2007 platforms.

Microsoft's Office Communication Server 2007 does not include call queuing, (it is believed call queuing shall be included in R2 of OCS 2007 - but how it will function or what it will offer is not yet clear).

Asterisk can provide call queuing - but Asterisk is not aware of the presence of a user in OCS. So how do we control Asterisk to pass calls to OCS users based on their presence?

Tools needed:
- Microsoft's C# - there is a free version of C# called the Express Version which maybe suitable for this - Visual Studio Professional 2005 or later works just as well.
- Asterisk.NET (1.4.0.1) - http://sourceforge.net/projects/asterisk-dotnet
- Microsoft Office Communicator 2007 SDK 
- Office Communicator Presence Controls

You can 'tweak' the example app provided with Asterisk.NET & demonstrate this can be done.

"Preparing your Persona"
Install the "Office Communicator Presence Controls".
Now open "Program Files\ Microsoft Office Communicator 2007 Presence Controls".

Here you will find a readme.doc. Pages 7 & 8 tells you how to compile the managed control - which will produce "PresenceControls.dll".

Remember where this dll is - we'll need it later. Close this project.

"Check Asterisk.NET works with your Asterisk server"

On your Asterisk server, edit your extensions.conf file and add these 2 lines in any context you wish to use:
exten => 200,1,agi(agi://<ip address of dev PC in here>/customivr)
exten => 200,2,Hangup()

Open Asterisk.NET in Visual Studio - once loaded - you should see two projects inside it:

Asterisk.NET & Asterisk.NET.Test

Under the Asterisk.NET.Test project - view the code in "Program.cs". Update the IP address & login credentials to match your Asterisk server.

From the Solution Explorer panel, right-click on the Asterisk.2005 solution - and choose Rebuild. When it finishes - navigate in Windows to the "bin\Release" subfolder under the Asterisk.NET folder. Run the Asterisk.NET.Test.exe

You should see a command window running the test app. You should now be able to use a softphone to connect to Asterisk - and dial 200. If you are watching the Asterisk console - you should be able to see Asterisk receiving instructions from the test app running on your desktop.

"Amend the Asterisk.NET test app to react to OCS presence"

Return to the Asterisk.NET solution in Visual Studio. Right click on Asterisk.NET.Test - choose Add...- and from the sub-menu, choose Windows Form... I shall leave the name of the form as default - Form1.cs

Look at Form1.cs in the Designer View (not code view). Right-click anywhere in the toolbox panel & select "Choose Items..." You should see this window...



Press "Browse...". Now find the "PresenceControls.dll" from earlier. This will add the following two controls to the toolbox.


The "Persona" control is to monitor one user, and the "PersonaList" monitors multiple users.

Drag a "Persona" control onto your form. Now change the "Modifiers" property of the "Persona1" control to Public.

Now change to the code view for Form1.cs. Immediately after the InitalizeComponent line, I added a line to assign a user to the persona control - as below:

InitializeComponent();
persona1.SipUri = "<username>@<domainname.com>";

Once assigned a SipUri - the Persona control will register with the OCS server for that user - and continue to receive updates from the OCS server whenever the presence changes for that user.

You do have to have Office Communicator installed for the persona control to work - but the assigned user can be any user that you can detect presence for. Basically - if you have the permissions to add a user to your contacts in the Office Communicator (OC) client - you can monitor the presence of that user here.

Move to the code view of Program.cs - and change line 29 to exclude the checkManagerAPI(); command - like this below:

// checkManagerAPI();
checkFastAGI();

Move to the code view of CustomIVR.cs - here is the code controlling Asterisk when you dial the extension numbered 200.

Approx line 40 - after the answer command, add a new line to create an instance of the form1 we created.

Form1 testform = new Form1();

The form contains the persona control for our user.

We then use the persona control within the IVR code to control the call flow based on the presence of that user. "TextStatus" from the Persona control will give us a text response of the status of that user.

Use this line to display to the console the status of the user:

Console.WriteLine(testform.persona1.TextStatus);

Then use this code anywhere within the IVR code to control call flow based on the presence of the user in OCS.

if (testform.persona1.TextStatus == "Busy" || testform.persona1.TextStatus == "On the Phone")
{
<runs some code in here if the user is busy or one the phone>
}

Note that within OCS the "On the Phone" status is used when a user is on the phone (doh!) - but from an OC client it appears their status is set to "Busy".

If a user is not logged in to OCS - the presence is set to "Unknown". It also shows the text from the four OC client custom presence states if you have used them.

Rebuild the Asterisk.2005 solution - and run the Asterisk.NET.Test.exe again. Now when you call 200 from a Asterisk connected softphone - you should see the presence of the OCS user you are monitoring appear on the console of the test app each time the IVR menu 'loops'.

To help with troubleshooting - this is the contents of the "Release" directory when finshed.


Note: Make sure you have you need RTMPLTFM.dll & Uccp.dll in the working directory.

Happy OCS 2007 presence integration with Asterisk! If you try it, let me know how it goes.

Tags: Asterisk, Asterisk.NET, microsoft, OCS, OCS 2007, Office Communicator, presence, Visual Studio, voip

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• Office Communications Server 2007 R2 Rumors - May 15, 2008
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• Microsoft & Aspect Software Partner on Unified Communications - Mar 18, 2008

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July 21, 2008 08:19 PM

Please add links to the menu on the left for the other projects, Asterisk is not the only Open Source VoIP app. FreeSWITCH http://www.freeswitch.org sipX http://sipx-wiki.calivia.com/index.php/SipX#sipX_-_The_SIP_PBX_for_Linux CallWeaver http://www.callweaver.org YATE http://yate.null.ro/pmwiki/ Bayonne http://www.gnu.org/software/bayonne/ OpenSER http://www.openser.org/
anthm () at 2008-07-18 18:44 GMT

July 18, 2008 06:44 PM

Isn't there something a bit more recent available? This book is 5 years old by now...
evert (Evert Meulie) at 2008-07-17 07:08 GMT

July 17, 2008 07:08 AM

XO Communications is one of the largest Competitive Local Exchange Carrier (CLEC) in the country. XO provides voice, data and IP services to businesses and other telecommunications companies in 75 metropolitan markets across the United States. XO Communications offers businesses advanced IP and networking solutions to maximize performance and simplify management of their networks. They also offer SIP services, hosted IP-PBX functionality (the XO One iPBX 50 powered by Avaya IP Office), VoIP origination/termination, SIP trunking, and more.

One of their flagship products, the XO IP VPN is a network-based Wide Area Network (WAN) solution delivered over the XO nationwide IP network. It's a solution aptly suited to businesses with multiple sites. The IP VPN is an advanced network allowing for faster application deployment, lower network operating costs, robust Class of Service (CoS) capabilities, and more access options than traditional WAN services.

Further, the data is segregated from other customers and the public internet. Importantly, XO offers competitive Service Level Agreements (SLAs) on packet loss and jitter.XO provides secure communications for multi-site networks, delivered over the XO private, MPLS-enabled IP backbone.

I find it interesting that XO uses IP MPLS services rather than Ethernet VPLS services. IP MPLS uses multiprotocol label switching (MPLS) over a public or private Internet connection. The advantage is that it an support any-to-any connectivity with CoS/QoS. MPLS's class of service (CoS) tagging and prioritization of network traffic, makes it easy to specify which applications should have priority. Packet classification makes an MPLS network especially important to customers that need to ensure the performance of low-latency applications such as VoIP. Additionally, MPLS carriers can offer tiered prices for each CoS tier. The disadvantage of MPLS is that it's a costly transition and complex operation.

Ethernet VPLS services on the other hand use virtual private LAN service (VPLS) over a carrier Ethernet network to provide a WAN that is configured like a LAN. The advantage is that it's simple, supports any-to-any connectivity with CoS/QoS and has lower total cost of ownership (TCO) than MPLS. But if XO built out their IP MPLS infrastructure already, it probably doesn't make sense to switch to Ethernet VPLS.

In any event, XO utilizes a nationwide OC-192 Tier 1 network along with a sizable fiber optic network, including an 18,000 route-mile inter-city network and more than 9,000 route-miles within 40 major metropolitan markets. XO claims they carry more than 15 billion minutes of VoIP traffic across its network each year.

The XO MPLS IP VPN service is a network-based Wide Area Network (WAN) solution delivered over the XO nationwide IP network. The IP VPN offers multi-site businesses more bandwidth for the dollar, faster application deployment, lower network operating costs, robust Class of Service.

Finally, XO has an IP VPN channel on TMCnet worth checking out with some good resources on IP VPNs. It includes a link for signing up for an IP VPN newsletter, news on various IP VPN industry happenings, IP VPN whitepapers, webcasts, and customer profiles. Go check it out.

Tags: CLEC, CoS, Ethernet, ip, IP MPLS, IP VPN, MPLS, QoS, TCO, VoIP, VPLS, XO, xo communications

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July 16, 2008 03:40 PM

Hi, our customer need to record some non-crypto (rtp) and crypto (srtp) phone call in a VoIP asterisk network; Is there a solution for this issue? Thanks
seamaster (Massimiliano) at 2008-07-16 14:23 GMT

July 16, 2008 02:23 PM

The url is now http://www.cats-muvva.net/software/asterisk/. My domain expired and some b*stards are squatting on the old one so I cannot re-register it. So, soddem!
catsmuvva (Nicole) at 2008-07-16 13:09 GMT

July 16, 2008 01:09 PM

Kristian Kielhofner has posted details of a new AstLinux install script to test.

July 16, 2008 03:25 AM

John Todd from Digium has posted asking people if they would like to go on a balloon trip

July 16, 2008 02:12 AM

Ronald Lewis has posted a preview of his howto guide for running Asterisk on Amazon EC2.

July 16, 2008 01:53 AM

Darren Sessions has posted details of a release of app_swift for Asterisk 1.2.x and a few additions.

July 16, 2008 01:32 AM

So it escaped me at first but this SPA-24DDTR card will give you a 24Channel trunk but not CallerID so I want to point out PRTC (PRI) is needed to do CallerID
MacAries (Rob MacNaughton) at 2008-07-15 14:53 GMT

July 15, 2008 02:53 PM

Snom, the German VoIP handset manufacturer today announced that their wideband enabling technology called KlarVOICE is now available in the North American market.

KlarVOICE, which can be adapted to all snom VoIP telephones, allows the capture of more than twice the spectrum of voice frequencies captured by standard phones for a high-fidelity VoIP experience. Key features of snom KlarVOICE powered handsets include full support for the G.722 codec and the doubling of the sample rate, providing an effective pass-band of 50 to 7,000 Hz as opposed to the 200-3300 Hz. The snom klarVOICE handset works with the codec G.722.2 which is able to shrink the bit rate of the voice channel down to 12.65 kbps.

snom's SIP-based phones are often used in Asterisk and other SIP-compatible phone systems.

Pricing and Availability
The snom klarVOICE handset, which can be adapted to any existing snom 3xx series VoIP phone (snom 300, 320, 360, 370) using snom's latest firmware release (Version7.1.33), is available for an MSRP of US$32.50.

Tags: handset, KlarVOICE, snom, snom 3xx, wideband, wideband codecs, wideband telephony

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July 14, 2008 08:55 PM

Hi, Is this still open for Administration? Thanks. Raymond
mondster () at 2008-07-14 15:45 GMT

July 14, 2008 03:45 PM

ITConnection.ru Is Authorized Partner Digium ® on training Asterisk ® (Asterisk ® Training) to Russia. We offer a full spectrum of rates Asterisk ® Training and world-wide recognized certification dCAP (Digium Certified Asterisk Professional). ITConnection.ru Together with Digium ® offers training Asterisk in Saint-Petersburg in Russian. Asterisk Bootcamp - a 5-day's rate which includes all key questions on installation, configurations and to administration Asterisk ®. Each employment will consist of lectures and practical works. Upon termination of training you become expert Asterisk and at will can hand over the test on dCAP. Cost of course Asterisk Bootcamp - 87438,00 roubles, including of 18 %. Also you can pass examination for reception of the Certificate dCAP. Cost of examination - 8743,80 roubles, including of 18 %. It is recommended to pass all over again a course, then to pass examination on dCAP. It is possible to take over examination for reception of the Certificate dCAP, not passing course Asterisk Bootcamp. ATTENTION: passage of course Asterisk Bootcamp does not guarantee passing an examination on dCAP. Examination on dCAP consists of written 150 questions rather Asterisk and with it of the connected technologies, and practical laboratory work in which you should adjust PBX under the set specification. It is necessary to give a minimum of 75 % of right answers for passage of examination and reception of the certificate dCAP. Preliminary record of listeners on course Asterisk Bootcamp is conducted. Applications we ask to send on training@itconnection.ru. Course will begin on August, 25th this year in St.-Petersburg. To reserve a place on an accessible course to you it is necessary to bring 100 % an advance payment of cost of a rate. The place will not be reserved without payment. Day of payment considers day of receipt of money resources on our settlement account. For successful passing an examination on dCAP it is recommended to pass curriculum Asterisk Bootcamp, to read through book O'Reilly " Asterisk: The Future of Telephony " which is accessible in on-line, PDF it is possible download free of charge, to study configuration files Asterisk. Samples of configuration files are accessible in Asterisk or on http: // voip-info.org. Site Asterisk (http: // www.digium.com/en/training) too a good source of the information.
kl007 () at 2008-07-14 09:48 GMT

July 14, 2008 09:48 AM

hi, i've been trying for months and reading this site thoroughly but I couldn't find an answer so far. Can you call an asterisk extension that dials somebody, hangup after a conversation and send the outbound call to a different context for IVR? Thanx!
jhvdp (jhvdp) at 2008-07-13 18:42 GMT

July 13, 2008 06:42 PM

I used the instructions, but the had some slight differences. 1) I had to select PRI as the line type before I could select Magix-NTWK as my switchtype. Note is was also Magix-NTWK and not Legend-NTWK 2) It did not create a 890 Pool for me. This is possibly because I have 8 defined as a Trunk/Pool selector. I ended up putting all of the trunks into Pool 592. 3) I am trying to figure out how to let calls coming from Asterisk to the outside PSTN allowed. 4) Still trying to figure out how to get Caller ID to be picked up on the Magix. It works from the Magix to Asterisk, but not back. If anyone has this working, I would love to know what it took.
jcrew77 () at 2008-07-13 05:34 GMT

July 13, 2008 05:34 AM

Can anyone offer information on the availability of this command as it is not showing up in my 1.4.21.1 installation compiled with PGSQL paths specified. I need to port an app written with the MYSQL() command (hence the Realtime/ODBC solution is not practical). Is this reference to a command that has been removed or will be added?? Happy to update the cmd_PGSQL page if someone can advise (and I can get it working!)
ijh () at 2008-07-12 11:07 GMT

July 12, 2008 11:07 AM

July 11, 2008 08:13 PM

Just two days ago I highlighted the connections between Adtran and Digium. They're both located in Huntsville, Alabama, Mark Spencer worked at Adtran, their current CEO Danny Windham came from Adtran, etc.

Well, today Rich writes about Leslie Conway, who also comes from Adtran, joining the Digium team, thus adding to the Adtran/Digium connection.

Rich writes:

Over the years, Mark Spencer, the founder of the company transitioned from CEO to CTO and during the transition Digium tapped Adtran for talent as needed. Adtran was an early investor in Digium and this may have been one of the smartest investment decisions a company can make.

Well said, Rich...

Tags: Asterisk, Digium, Leslie Conway, Mark Spencer, VoIP

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• Hulk Smash Asterisk 1.6! - Apr 16, 2008

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July 11, 2008 03:18 PM

If the trunkfreq parameter is 30ms Asterisk will send Meta Trunk Frame every 30ms. If you use a sample period of 20ms you will not have any advantages. If you use 30ms, you should use the bandwith better. It depends on the number of channels that you have. There are limiters in max channel number in an IAX trunk. Is the trunkmaxsize parameter. Gianrico Fichera
gianrico () at 2008-07-10 20:43 GMT

July 10, 2008 08:43 PM

According to eFluxMedia, Comcast is "working on rehabilitating its name and implementing reasonable management techniques through a new partnership with VoIP service provider Vonage." Vonage and Comcast said they will work on ensuring adequate management techniques to avoid network congestion to ensure high quality VoIP services.

I should point out that Comcast tarnished their own reputation when they intentionally degraded P2P traffic, particularly Bittorrent, a heavy bandwidth application. According to a 36 page thread on the Vonage Forums that dates back to 2006, Comcast was accused of degrading Vonage's voice over IP quality intentionally. Comcat has denied these charges, but many Comcast users that have Vonage have had issues.

Whether conspiracy or not, now Vonage and Comcast stated they will have a "direct line of communications" between their network operations centers to resolve customer issues. Umm, so they couldn't talk to each other easily before, so now they need a special 'bat phone' direct hotline?

According to the Free Press, Marvin Ammori, general counsel of Free Press and author of the complaint, issued the following statement:

"We are baffled as to why it was necessary for Vonage to strike a network management agreement with Comcast to guarantee that their services are not degraded or blocked. Such anti-competitive, anti-consumer practices are already against the law. And beyond that, Comcast has been on the record as saying that they do nothing to deter their customers' use of VoIP.

"This announcement calls into question the company's honesty about its treatment of competing services. Was Comcast degrading Vonage's VoIP service before this announcement? And are they continuing to degrade other services that compete with their products? That these questions remain unanswered by today's announcement is cause for great concern. This collaboration should do nothing to deter the FCC from investigating and stopping Comcast's blocking other Internet services."

The partnership with Vonage is supposedly part of Comcast's commitment to move to a protocol-agnostic network management approach by the end of 2008. Comcast has announced collaboration with Pando Networks for a "P2P Bill of Rights and Responsibilities" (BRR) and participation in the P4P Working Group organized by the Distributed Computing Industry Association.

It all sounds well & good, but we'll see if Comcast lives up to their word to play fair and not mess with IP packets. I for one am not holding my breath.

Tags: bat phone, Bittorrent, Comcast, P2P, VoIP, Vonage

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July 10, 2008 05:52 PM

You remember the scene in Jurassic Park where the there was no power or working phones on the island so they head to the control room? Lex Murphy, the young hacker girl, gets on a computer, works some computer wizardry and gets the power and phones back online and is able boot up the door locks and other systems - though its not enough to keep the raptors from getting to them.

Well, there's a funny Youtube parody video that solves the apparent "IP phone" outage simply by restarting the Cisco Manager services.

Ironically, the girl says it's a UNIX system, while Call Manager is Windows-based.

Preview snapshot:


Click Play Below

Tags: CallManager, Cisco, Jurassic Park, Lex Murphy, raptors, VoIP

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July 10, 2008 05:10 PM

Hello, Is it possible to affect a specific musiconhold, not default, on an external call? If yes, how can I do it? Thanks
jpawlak () at 2008-07-10 07:33 GMT

July 10, 2008 07:33 AM

I followed the installation instructions and got up the point of installing the amd_conf.c file with astxs, unfortunately, I can't seem to find astxs anywhere on my system (I do have the source code installed and asterisk has been working for sometime now). Any tricks/tips would be appreciated.
flajeff () at 2008-07-09 12:42 GMT

July 09, 2008 12:42 PM

I just happened to be checking out our firewall logs and noticed traffic coming from 88.160.222.185. Curious, I did a whois and figured out it was coming from http://www.proxad.net/ which redirects to http://www.free.fr/adsl/.

The first thing I noticed other than the fact that website is in French,-- which I can't read -- is that they offer Internet + Telephone + Television for 29.99 €/month. I was able to figure that out since apparently Internet, Telephone, and Television don't translate at all in French. They're the same words except for some accent letters, as seen here from a website screen grab I did:

What I can gather is that they offer 250 channels, ADSL (ADSL2+?), Freebox HD receiver, unlimited phone calls to 70 destinations, and even a WiFi-MIMO (multiple-input and multiple-output) router. MIMO (pronounced mee-moh or my-moh), is the use of multiple antennas at both the transmitter and receiver to improve communication performance. Their Internet speed says 28 Mega- whatever that means. 28Mbit/s? Although, ADSL2+, which is faster than ADSL, maxes out at 24Mbits/s. Hmmm. Are the French inventing some proprietary ADSL spec that is faster?

Also, if I read their website correctly, they give you 10GB of storage space - mostly likely talking about the Freebox receiver. Seems a bit low to me if doing any sort of Tivo-like functionality (pause Live TV, recordings, etc.) The 29.99 €/month translates into $46.99 U.S. dollars which seems like a pretty sweet deal for a triple-play package!

Any French readers want to translate exactly what this products' feature-set is? Post a comment...

Update: 5 min after post Found some more info on Wikipedia

The box, designed by Free, uses a 32 bits RC32355 processor and is managed by an operating system using a derivative of the Linux kernel. It has many interfaces:

• An Ethernet port 10/100 Mbit/s full/half duplex;
• A USB2 port;
• An HDMI port;
• An RJ11 jack for the ADSL connection;
• An RJ11 jack for phone equipment (two jacks on versions 1 & 2 but only one active);
• A SCART (Péritel) socket
• An digital audio output RCA, or optical SPDIF starting from version 3;
• An extension port of the Serial ATA standard on versions 3 and 4 and Parallel ATA standard on versions 1 and 2;
• A host USB port on version 4;

One cool feature is that it supports a Videolan client in order to get the movies (in any format read by VLC) stored on the computer and watchable on TV through a playlist selector using Freeplayer. See my VideoLan post for more on this free streaming client.

Freebox is indeed an ADSL2+ modem that the French ISP called Free provides to its ADSL subscribers at a cost of around 190 Euros.

It can not only be uses as a high-end wireless modem (802.11g MIMO), but it also enbles Free to offer value-add services such as HD television (1080p), video recording with timeshifting capabilities, digital radio and analog telephony via one or more RJ11 ports.

Tags: Freebox, Freebox HD, IPTV, television, triple play, TV, VoIP

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July 08, 2008 07:50 PM

Sheesh, what is this, VoIP Contest Day? I just blogged this morning about a SIP/IMS contest sponsored by mobilkom austria and now BroadSoft is launching a voice/VoIP mashup contest. They wrote me to say, "We're looking for mashups that will revolutionize telephony services from today's one-size-fits-all offerings to the Long Tail internet approach where users decide what services are valuable. Winners will receive cash, plus the opportunity to demo their mashup live in front of hundreds of service provider executives at BroadSoft Connections."

Full news after the jump...

Continue reading BroadSoft announces Xtended Voice/VoIP Mashup Contest for Cash...

Tags: BroadSoft, contest, mashup, mobilkom austria, VoIP, Xtended

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