The beginning of the end for analog landline phone lines?
If the Ohio house has its way, local phone companies in Ohio will be able to stop providing landline phone service to under performing areas.
This comes at a time where VoIP is cutting into profits of local phone service providers, VoIP providers can provide service to anyone without building and maintaining expensive infrastructures, because they are relying on existing infrastructures that customers already have.
Same is true as cell phone carriers are offering unlimited plans at reasonable prices causing lots of people to forgo landlines altogether, although lots of rural areas don’t have full cell phone coverage therefore this bill is angering lots of residents.
Ohio University is in the process of moving over their 10,000 landline phone lines to VoIP, so they won’t be affected by this bill.
If this is going to become a trend and other states will follow suit remains to be seen, but eventually all phone service will be digital and the local phone companies that already have the infrastructure built and need to keep it running in order to provide internet service will switch to VoIP in order to cut costs.
Phone companies already provide VoIP products, like FiOS Digital Voice from Verizon, but it runs on FiOS which is not available all over, especially in the rural areas which naturally would be the first areas that phone companies would want to terminate local phone service, a previous attempt by Verizon to provide VoIP under the “VoiceWing” brand has failed.
Will cellular roaming soon be a thing of the past?
the answer in short, probably partially.
For years I’ve been saying that eventually cell phone service will be limited to data, voice will be carried over the internet via VOIP, the exciting thing about this is that you can travel anywhere and buy a local SIM for data, and talk on the phone as much as you want as if you’re in your home country.
This technology has been available for a while now using VOIP applications that are either built into the OS or third party apps, I’ve had mixed success with VOIP over 3g, some areas the voice was as clear as a landline, and in other areas there were lots of issues with latency, jitter, bad 3g coverage or carriers blocking VOIP traffic.
But now with 4G emerging rapidly in more and more markets, VOIP over 4G is slowly becoming a reality, carriers are starting to roll out VoLTE (Voice Over LTE) which is the cellular carriers answer to VOIP, this technology has multiple advantages over SIP notably that it can handle echo cancellation on the network side, thus not relying on the handset which can be a great advantage if the handset does not have enough muscle to handle the tedious task of echo cancellation, carriers also have control over their network therefore can offer QOS for voice traffic.
VoLTE will probably not be of an advantage for the traveler, because carriers will probably not be willing to give up on their income from roaming minutes, also because you will need to use data in order to use VoLTE and international data roaming is still very expensive.
But with smart phones becoming faster and faster, with quad core cpu’s found even in entry level phones the advantages of network side echo cancellation might not be an advantage at all in the near future, and with 4g networks speed QOS might not be that important and echo will be greatly reduced, sip alternatives will probably become more and more attractive for the traveler.
I haven’t tried yet VOIP over 4g, but a associate of mine recently traveled to a foreign country, he bought a 4g hotspot (like a MIFI) and he used his tablet to make VOIP phone calls, he claimed it was as clear as his landline phone.
With prices for unlimited minutes falling every day using a third party sip provider might not be a viable alternative for the regular user, most SIP carriers dont provide SMS which is another drawback, but for a traveler nothing can be more exciting than to pick up the phone at the Eiffel tower and not rushing to get off the phone because every minute on the phone is more expensive then a minute flying on the airplane.
So until carriers will stop charging for roaming, the best option would probably be to forward your cellular (and home) calls to a VOIP carrier, and use a VOIP enabled 4G phone with a local SIM, with a SIP provider that would let you set the outgoing caller id to your cell phones phone number.
How to create Custom Caller ID on Asterisk
For years I was ofering my customers custom caller id which changes the outgoing Caller ID according to the destination, its used mostly by companies who do business in multiple countries and they have DID’s in those countries and they want the outgoing caller id to display the local DID.
Recently I’ve seen several companies advertise this service as if they discovered a major breakthrough when really all it takes is one single database query to be called before the call is placed.
No AGI programming is required all thats needed is to compile Asterisk with the mysql module or the odbc module.
The way I do it is very simple, I created a table with 4 fields, (id, device, prefix, callerid) where device is the name of the sip peer, prefix is the prefix of the country and area code you wish to set the outgoing caller id, callerid is the caller id you want to send, here is the code to create such a table in MYSQL
create database customcid;
CREATE TABLE `customcid` (
`id` int(11) NOT NULL AUTO_INCREMENT,
`device` varchar(20) NOT NULL,
`prefix` varchar(10) NOT NULL,
`callerid` varchar(20) NOT NULL,
PRIMARY KEY (`id`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1;
Next I wrote a gosub macro which queries the Database and if it finds a caller id for the destination it will set the outbound caller id accordingly, here is my macro it can be rewriten to use ODBC, etc
[customcid]
exten => s,1,MYSQL(Connect connid localhost username password customcid)
exten => s,n,GotoIf($["${connid}" = ""]?error,1)
exten => s,n,MYSQL(Query resultid ${connid} SELECT prefix, callerid FROM customcid WHERE device = ‘${CHANNEL(peername)}’ and prefix = SUBSTRING(‘${ARG1}’,1,LENGTH(prefix)) AND LENGTH(SUBSTRING(‘${ARG1}’,1,LENGTH(prefix)))>0 ORDER BY LENGTH(prefix) DESC LIMIT 1)
exten => s,n,noop(${resultid})
exten => s,n(fetch),MYSQL(Fetch fetchid ${resultid} prefix callerid)
exten => s,n,noop(${prefix} ${callerid})
exten => s,n,MYSQL(Clear ${resultid})
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,set(CALLERID(num)=${callerid})
exten => s,n,return
exten => error,1,noop(Database Connection Error)
exten => error,n,return
Next you will need to call the macro before completing the call
exten => _X.,1,gosub(customcid,s,1(${EXTEN}))
exten => _X.,n,noop(Regular routine to complete call
Now all thats left to do is to populate the table, it is as extremely flexible, for example you can have one number for USA and another number for USA area code 212, all you need to do is enter 1 in the prefix, and in the next record enter 1212 and the query will chose the right one, you can even customize it down to a certain number, for example if you want to set a certain outbound caller id when you call 1-212-555-1212 then enter 12125551212 in the prefix field.
Why you don’t become a VOIP provider by simply purchasing a dedicated server
Over the years that I’ve been doing VOIP consulting, I’ve come across more and more companies who consider themselves VOIP providers, who don’t have anything but a single Asterisk server hosted in a data-center, most of these companies are run by opportunists who hear over and over that VOIP is the future and they want to jump on the bandwagon, some are more technical savvy and some are less, but the common denominator is that none of them are prepared for disaster, and as we know we need to plan not because “if” disaster happens, but for “when” disaster happens, sooner then later it will hapen.
One of my clients even had his server located in his office on a Cable internet connection, and when a backhoe 2 blocks away cut the cable for the entire neighborhood, he quickly brought the server home to connect to his DSL connection at home, this guy was advertising in his local paper, and had no idea what he was doing.
Here is why becoming a VOIP provider is not as easy as the do it yourself websites might want you to think, and why planning is crucial for the small VOIP provider.
1) Redundancy: You need multiple layers of redundancy to have a reliable network, Godaddys incident this week is a good example why you cant rely on any single source even on the largest registrar, in order to have reliable redundancy you need to host servers in multiple networks, and you need to implement proper failover methods, like heartbeat, dns failover, etc for every service including SIP Registry, media and database etc., also when relying on DNS you will need more then one DNS provider so if the dns provider fails like we just had with Godaddy you will still be up.
2) Load Balancing: you need to make sure that your servers can handle all your users at peak, therefore you need to implement proper load balancing and add more servers as required, sometimes your redundancy solutions will also provide load balancing, therefore proper planning is crucial.
3) Routes: You need high quality routes, because if you have the best equipment with lousy routes your customers will not be happy, your customers expect the best quality and every little quality issue like echo, jitter, latency of packet loss will make very unhappy customers, you will need multiple carriers just in case one carrier cannot handle the call, simply buying routes is not enough, if dealing with lots of destinations finding the right route for every destination can be a nightmare, some companies have entire teams of specialists to manage routes, simply using a LCR program is not going to do too much good when a route needs to be removed for a certain destination.
4) Customer support: Problems happen often when using VOIP, most of the times its related to the internet connection, but even when its not your issue, you will need to make the diagnosis and guide the customer on how to handle the issue with his ISP, ISP’s will often throw the ball back in your park just because they can, lots of times their techs have no idea whats jitter etc, they might tell you as long as you can browse on the internet there is nothing they can help you, in the end the customer will come back to you, and as much as you try to explain to him that IP traffic is very sensitive because we need it in real time and cant rely on packet re-submission he will still blame you, handling problems like this can be very time consuming, and require techs who really understand sip and can run a trace on sip and RTP traffic, simply knowing how to set up Asterisk is not going to suffice.
5) Techs that know your system: There will always be bugs and glitches that need to be fixed, there will always be that new feature that you need to add to your system in order to stay competitive, sometimes things have to be fixed right away, and relying on one person working part time is going to put your business down the drain.
6) All the rest: Like every other business you need the proper sales team, the right financial adviser etc, it all comes down to money, and if skipped be ready to be added to the long list of companies buried in the VOIP graveyard.
So to put it in a few words you need Planning Planning Planning and more Planning, and of course Money Money and Money
Asterisk 10.8.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.8.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 10.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
— AST-2012-012: Resolve AMI User Unauthorized Shell Access through
ExternalIVR
(Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research)
— AST-2012-013: Resolve ACL rules being ignored during calls by
some IAX2 peers
(Closes issue ASTERISK-20186. Reported by Alan Frisch)
— Handle extremely out of order RFC 2833 DTMF
(Closes issue ASTERISK-18404. Reported by Stephane Chazelas)
— Resolve severe memory leak in CEL logging modules.
(Closes issue AST-916. Reported by Thomas Arimont)
— Only re-create an SRTP session when needed
(Issue ASTERISK-20194. Reported by Nicolo Mazzon)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0
Thank you for your continued support of Asterisk!
Godaddy CEO Addresses Sept. 10 Service Outage
Go Daddy Customers and Community,
We owe you a big apology for the intermittent service outages we experienced on September 10th that may have impacted your website and your interaction with GoDaddy.com.
The service outage was due to a series of internal network events that corrupted router data tables. Once the issues were identified, we took corrective actions to restore services for our customers and GoDaddy.com. We have implemented measures to prevent this from occurring again.
At no time was any sensitive customer information, such as credit card data, passwords or names and addresses, compromised.
Throughout our history, we have provided 99.999% uptime in our DNS infrastructure. This is the level of performance our customers have come to expect from us and that we expect from ourselves. We pride ourselves on providing world-class service — through our products, our site experience and customer care.
We have let our customers down and we know it. I cannot express how sorry I am to those of you who were inconvenienced. We will learn from this.
I’d like to express my profound gratitude to all our customers. We are thankful for your straightforward feedback and the confidence you have shown in us.
In appreciation, we will reach out to affected customers in the coming days with a good faith gesture that acknowledges the disruption. We are grateful for your continued loyalty and support.
If you have any questions or require further assistance, please call us at 1-480-505-8877.
Sincerely,
Scott Wagner
Go Daddy CEO
AnonymousOwn3r seems to be a fake
AnonymousOwn3r who claimed to be the Godaddy hacker, posted on twitter that he has the source code that runs Godaddy to prove that he is the Godaddy hacker, but a simple google search on the file names revealed that this code was submitted as opensource by ttpython project in 2010.
This just makes me feel that this so called hacker is just a attention seeker who decided to take advantage of the situation.
Godaddy also posted today the following:
The service outage was not caused by external influences. It was not a “hack” and it was not a denial of service attack (DDoS). We have determined the service outage was due to a series of internal network events that corrupted router data tables. Once the issues were identified, we took corrective actions to restore services for our customers and GoDaddy.com. We have implemented measures to prevent this from occurring again.
At no time was any customer data at risk or were any of our systems compromised.
Here is a link to the Godaddy statement http://www.godaddy.com/newscenter/release-view.aspx?news_item_id=410
Here is a link to AnonymousOwn3r’s Godaddy code taken from his Twitter page http://pastebin.com/WMb4Qg7H
here is the link to the opensource project https://code.google.com/p/ttpython/source/browse/#svn%2Ftrunk%2Fgodaddy
Godaddy DNS goes down bringing down with it some VOIP providers.
Godaddy DNS has been up and down for a few hours now(Sep 10th 2012) their own website and phone support is up and down too, Godaddy is one of the biggest registrars, and naturally lots of web sites use their free DNS service, some VOIP providers who are using godaddy’s DNS are unreachable too.
Lesson learned VOIP providers invest lots of money in redundency and sometimes the most overlooked point is the DNS server, the chain is only as strong as its weakest link, so go out and pay a few bucks and get a backup DNS provider.
UPDATE: Claims by AnonymousOwn3r that he single handily brought down Godaddy seems to be false, read more about it http://sipguy.com/anonymousown3r-seems-to-be-a-fake/
Why is the Asterisk queue app so neglected?
Although Asterisk was originally built as a PBX, VOIP providers have realized a long time ago that when combined with proper load balancing software like OpenSIPS/Kamailio it can be as powerful as software costing hundreds of thousands of dollars.
Over the years more and more companies have realized the power of Asterisk, and it has become a major player in almost any phone related business, like call centers, online ordering etc.
Asterisk has become more and more friendly to load balancing and multi tenants with the introduction of Asterisk Real Time, ODBC voice-mail storage etc. where multiple asterisk servers can work independently of the other and still share all the needed info, Digium has also added multiple parking lots which really helped companies providing hosted PBX service.
There is one major issue which can be a killer for companies providing hosted pbx service or large call centers who rely on Asterisk for their incoming calls, its the “queue” app, the queue app cannot share states with other servers, therefore all callers on a queue need to be on same server which really makes it impossible to use in large installations.
I know there are workarounds like Vicidial but lots of companies don’t allow third party apps for multiple reasons mainly because of reliability and support reasons, it can also cause major headaches when upgrading Asterisk versions.
Therefore I’m really surprised that the app which is mostly used in big installations like call centers has been so neglected, its not a major deal to make it database aware, I understand that there are other details which would have to be worked out, for example agents might not be registered to same server, sometimes they are registered to a OpenSIPS server etc. and then asterisk does not know the state of the agent but this can also be a easy fix by using a shared database, I think that this is long past due.
T-Mobile Launches CleverConnect, A Bobsled-Style VoIP Service For Europe
T-Mobile today becomes the latest operator to leverage the popularity of free internet phone calls and texts in hopes of luring in more users around its brand. CleverConnect borrows from services like Skype and Rebtel with an offer of free VoIP calls and text to those who download and use the app, which is now live in the App Store and in the Google Play Android store.
Similar to Bobsled in the UK and services like TuMe from Telefonica, CleverConnect is aimed at bringing in users that are not necessarily already part of T-Mobile’s service. Unlike Bobsled, the service is available in multiple regions. Although the app is already in the App Store you need an invite code to use it for now, and it looks like you need to be a T-Mobile subscriber to set off that chain. TechCrunch understands, however, that those who get invited can bring in another three users to trial the service, regardless of country or carrier.
As with those other apps, users will need either WiFi or 3G to make the service work.
It has been developed in connection with a Danish mobile VoIP startup called Vopium. For those interested in mobile VoIP, this is a name you should watch — the company started first with its own B2C mobile VoIP services but now is out making deals and developing services with operators — T-Mobile’s CleverConnect being only the first, apparently. That’s filling a gap at a time when companies like Skype are not giving themselves over in similar white-label arrangements.
While users the apps will be able to make free phone calls when calling each other, the service will also offer users flat-rates for other calls as well. Those will be charged at 2p per minute in the UK and 5p per minute for the U.S. and Canada. 100 minute packages will go for $4.99, the site says.
Since T-Mobile launched Bobsled in the U.S., first as a Facebook app then as a standalone service, it’s been unclear how much traction the service has had.