A SIP trunk is the external component of an in-house IP PBX. The PBX connects calls within an . . . → Read More: How to Cut Company Costs by Using a SIP Trunk]]> A SIP trunk to the rescue? SIP trunks save companies thousands of dollars each year through Internet-based telephony technology, and are especially useful for companies that have many extensions but a low volume of simultaneous calls.

A SIP trunk is the external component of an in-house IP PBX. The PBX connects calls within an office, extension to extension. The SIP trunk relays calls to and from external phone numbers (outside of the office PBX set of extensions). This means that an all-IP office can have full connectivity with callers with other providers, traditional or VoIP. And for the company, all calls are still cheap unlimited VoIP calls.

But why would you go with SIP instead of a hosted PBX? Many people love hosted PBXs because they are a hands-off business telecom system that allows a company to pay a slightly higher fee (about $20/extension each month) to get unlimited calling on each extension, and complete support and maintenance from their VoIP provider.

SIP trunks are a little cheaper (about $10 to $15 per outbound line) and for a business that is capable of handling their own telecom system, they are a great inexpensive alternative to hosted. With a SIP system, the customer does have to support and maintain their own in-house IP PBX, but they can customize their business telecom system so that they are only paying for what they need. That it, they don’t need to pay for unlimited calling for every extension. Instead, they only need to buy as many external lines as they need for outbound calling. So, if a company only expects ten outbound or inbound calls at any time, they can pay for ten SIP lines instead of for unlimited calling for all 100 office extensions.

A SIP trunk is a great 2013 investment for companies looking to save money, while upgrading the phone system.

1. A SIP Trunks Makes All Calls Cheap Internet Calls

SIP trunking saves a company money in two ways:

1. SIP trunking gives an awesome a cheap option for unlimited calling to all numbers in the US.
2. SIP trunks are cheap and customizable. Pay $10 for each line every month instead of $20 for each extension.

Because SIP trunks use VoIP, all outbound and inbound calls are unlimited. These calls are all translated into Internet calls, which are far cheaper than traditional phone calls that need to travel on physical analog circuits. SIP trunking also offers number portability so a company can keep their current phone numbers, so as to not confuse their customers.

2. SIP Trunks May Work with Legacy PBX Hardware and Your Current Phones

Companies that have already invested a good deal of money in their legacy PBX systems and do not want to upgrade to a hosted PBX can still use SIP trunks. SIP trunks work with both IP PBXs and some traditional PBXs.

However, if your PBX is old, you may need to get an adapter so that it can run with VoIP. If your PBX is too old, it probably can’t be IP enabled.

Analog Phones with AVA Adapters

SIP trunking also allows businesses to keep their current phones.. The companies simply need to buy AVAs (analog voice adapters) so the older analog phones can translate analog signals into Internet data.

However, many times businesses are better served by replacing their old systems with new IP systems. Many customers report better call quality with hardware built for IP data.

3. SIP trunks Consolidate Extensions

Traditional phone service runs off of expensive copper lines, and every time a company needs more room for concurrent calls they have to buy more circuits. SIP trunking uses Internet bandwidth for unlimited concurrent calls, and bandwidth is easy to increase or decrease. SIP trunking saves a company from paying an unlimited plan for each extension line, since they only have to pay for a limited amount of SIP trunks to manage the simultaneous phone calls.

How SIP Trunks Cut Down on Extensions

With a SIP trunk, businesses only need to figure out how many inbound and outbound simultaneous calls they can expect on a daily basis, and then buy an appropriate number of trunks. For instance, if a business has 30 extensions, but the company only expects about 10 simultaneous calls at any given time, then the business needs to only buy 10 SIP lines.

4. SIP Trunks are Managed In-House

SIP trunking could be considered an in-house system because it depends on a PBX that you own and operate yourself. Of course, you could run your PBX from a colocation center, but at any rate, you as the customer have a large say in how your system is run and maintained because you run and maintain it.

However, this would not be an ideal setup for a small business that is not capable of managing their own PBX. If you are a larger business, you have an IT staff, or your feel capable of choosing and setting up your PBX, you are a good candidate for SIP.

A SIP trunk allows a company to save money on extensions and save money on outbound and incoming calls. Companies don’t have to pay for unlimited use for each and every extension, especially when only a ratio of them are being used. With SIP trunks, businesses have an ally to to save money without sacrificing quality and current machinery.

Jennifer Cuellar is a writer for My VoIP Provider. Read more about business VoIP solutions here: www.myvoipprovider.com/en/Business_VoIP

Here is the sample conf for asterisk:

Please open jabber.conf (/etc/asterisk/jabber.conf) in your favourite editor and add following configuration:

jabber.conf

This is where you set your gmail/gtalk account info and will register you with the Google server.

[general] debug=yes autoprune=no . . . → Read More: How to integrate Asterisk setup with gtalk, yahoo etc for calling?]]> Asterisk can be integrated with Jabber xmpp  in order to  integrate gtalk, yahoo, etc.

Here is the sample conf for asterisk:

Please open jabber.conf (/etc/asterisk/jabber.conf) in your favourite editor and add following configuration:

jabber.conf

This is where you set your gmail/gtalk account info and will register you with the Google server.

[general]
debug=yes
autoprune=no
autoregister=no

[gtalk_account]
type=client
serverhost=talk.google.com
username=username@gmail.com/Talk
secret=*****
port=5222
usetls=yes
usesasl=yes
buddy=buddyusername@gmail.com
statusmessage=”This is an Asterisk server”
timeout=100

gtalk.conf

This is where the settings for the actual calls are made:

[general]
context=google-in
allowguest=yes

;
[guest]
disallow=all
allow=ulaw
context=google-in

[buddy]
username=buddyusername@gmail.com

disallow=all
allow=ulaw
context=google-in
connection=gtalk_account

extensions.conf

…
[google-in]
exten => s,1,NoOp( Call from Gtalk )
exten => s,n,Set(CALLERID(name)=”From Google Talk”)
exten => s,n,Dial(SIP/my_sip_phones)[google-out]
exten => 200,1,Dial(gtalk/gtalk_account/buddyusername@gmail.com)

For configuring registration trunk we have to add this line (format) under trunk configuration as

username:password@your.provider.com:5060

……..

[didforsale_2] type=peer host=209.216.15.70 nat=no canreinvite=no disallow=all allow=ulaw allow=g729 dtmfmode=rfc2833 insecure=very context=from-trunk After copying the above lines, save and reload Asterisk.

For this we need to create Ringgroup and add extensions. Then create context for incoming calls so that when someone calls the DID, it will go to the ring group.

[ext-did-0002]

include => ext-did-0002-custom

This will configure a voicemail system to email the received voice message (with attach file option) to myemail@gmail.com, when a call . . . → Read More: How to configure voicemail system in Asterisk?]]> VoiceMail is used to leave a message if someone is not answering your call. The configuration in Asterisk is done in /etc/asterisk/voicemail.conf. In order to configure voicemail, add following line:
1000 => myemail@gmail.com,,attach=no|saycid=no|envelope=no|delete=no

This will configure a voicemail system to email the received voice message (with attach file option) to myemail@gmail.com, when a call to ext 1000 is not picked.

Design and Navigation The design of the web pages and the structure . . . → Read More: Website Launch Announcement: DidforSale launches new site]]> Did For Sale is happy to announce the launch of new website! The website has been redesigned to improve user friendliness and appeal.
In addition to the changed design and layout of the pages, new functions have been implemented in this version.

Design and Navigation
The design of the web pages and the structure of information have been changed to improve overview and usability. The new design and colours now reflect the general DidforSale image and colour coding of our business areas.

Improved Product Pages
The presentation of the products in the Products section has been improved with more information and an optimized navigation structure.

Announcements
Clients can now register themselves for the newsletter. Amongst the new features, the site contains integrated social media buttons for Facebook, Twitter and LinkedIn to foster improved communication with our clients.

We hope you will enjoy our new site. If you have questions, comments or suggestions please send them to contact-info@didforsale.com