Any company that wants to build a Contact Centre without the hiring a company to manage the PBX, IVR or automated outbound calls.
Open source Contact Centre software based on Asterisk with PBX and predictive calling capabilities. Handles incoming and outgoing calls, incoming email and web site chat support.
Continue readingOriginal article: http://ow.ly/zfig302hJgz
VICIDIAL is one of the most used Open-Source Dialers world wide for call centers using VoIP to make calls all over the globe. VICIdial can handle anywhere between 2 to 300 or more agents at a time making thousands of calls per day with great capacity.
VICIBox Server is the authorized installation CD for the VICIdial Call Center Suite. It’s based on OpenSuSE server and will correctly install VICIdial Call Center Suite very easily.
It is recommended to have basic operating system skills when installing VICIBox but it is not necessary. If you follow the basic Vicidial Installation Manual you will than have a fully functioning VICIdial system.
Setting the Trunk in VICIDIAL or the Carrier, can sometimes be tricky or time consuming for most customers. We have set up a detailed informative configuration of the Vicidial predictive dialer to help you start making calls.
Click here for a full guide on How to setup your Trunk in VICIDIAL / GOAUTODIAL:
http://www.switch2voip.us/contact-support/how-to-setup-your-trunk-carrier-provider-in-vicidial-vicidialnow-goautodial-vicibox
To start making and receiving calls using Switch2Voip please verify that your Asterisk VoIP server is configured as follows.
We authenticate IP-PBX SIP Trunking traffic by:
IP Authentication (IP address) or Digest Authentication (account and SIP password)
After you decide which switch platform to use, you will need to establish a SIP trunk with our US proxy server sipusa.switch2voip.us and input your IP address into our portal or register your switch with us. Alternatively, if your switch is not in Central or North America, you can use one of our international POPs to reduce transit delays, for Europe and Asia point to 111.235.152.141
Our IP’s
Configure the switch to allow for traffic from Switch2Voip
For our US Server allow: 169.132.196.33 and 206.20.196.19 (for the CLI to work you have to setup both IP’s or use sipusa.switch2voip.us)
For our UK Server allow: 213.166.103.6
For our HK Server allow: 111.235.152.141
For our BR Server allow: 206.20.196.31 and 177.53.194.33
For our PE Server allow: 206.20.196.33 and 169.132.196.43
For our AR Server allow: 206.20.196.34 and 169.132.196.44
[Switch2Voip]
Peer Detail
username=
type=peer
secret=
progressinband=never
port=5060
nat=auto
insecure=very
ignoresdpversion=yes
host= sipusa.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=g729&g711
User Detail
username=
user=<>
type=user
port=5060
context=from-pstn
canreinvite=no
allow=g729&ulaw&alaw
Registration
register=>account:password@sipusa.switch2voip.us
The IP Authentication method is normally simpler to provision and should be used only when you have a static IP Address. It is also somewhat more secure since your SIP trunk can only be used from the IP Address you provide.
With an open source applications (such as Asterisk), you can setup your SIP trunk with IP Authentication as follows:
[out-1]
type=peer
port=5060
nat=auto
insecure=invite
ignoresdpversion=yes
host= sipusa.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
[in-1]
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=169.132.196.33
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
[in-2]
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=206.20.196.19
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
If you are using a web-based Asterisk PBX (like FreePBX), IP Authentication setup is slightly different:
In “Outgoing Settings”, name the section “out-1”
Then, in “Peer Detail”, enter the following:
type=peer
port=5060
nat=auto
insecure=invite
ignoresdpversion=yes
host= sipusa.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
In “Incoming Settings”, name the section “in-1” in “User Context”. Then, in “User Detail, enter the following:
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=169.132.196.33
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
After this has been completed, you will have to create a separate trunk. For the second trunk, name the outgoing “out-2” and again enter the following information:
type=peer
port=5060
nat=auto
insecure=invite
ignoresdpversion=yes
host= sipusa.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
Then, for the second trunk, name the incoming “in-2” and again enter the following information:
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=206.20.196.19
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
No registration string is required for IP Authentication.
Please make sure to configure your router/firewall to allow traffic from:
– sipusa.switch2voip.us for US
– 213.166.103.6 for UK
– 111.235.152.141 for Hong Kong
In addition, please allow all RTP traffic from any IP Address ports 20000-24000 UDP.
exten => _91.,1,Set(callerid(num)=+1XXXXXXXXXX)
exten => _91.,2,Set(callerid(ani)=Phone number)
exten => _91.,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _91.,5,Hangup
Make sure you change the prefix on your UK campaign to 8 and leave 9 for USA. Copy everything below this line and paste it on your dialer trunk configuration.
exten => _91.,1,Set(callerid(num)=+1XXXXXXXXXX)
exten => _91.,2,Set(callerid(ani)=Phone number)
exten => _91.,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _91.,5,Hangup
exten => _8011.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8011.,2,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _8011.,3,Hangup
A-Z Voip Termination for Call Centers.
Switch2Voip offers competitive, quality A–Z VoIP Termination for Call Centers We have been serving large business, call centers and residential VoIP customers for over 10 years in 55 countries. Our VoIP solution handles and regulates billions of calls each year with a 99.999% accuracy and certainty.
Read our next article: Switch2Voip.us offers competitive A-Z VoIP Termination for CallCenters
Switch2Voip has the best SIP termination and H.323 that every carrier has to give combined with great customer service, and with our pre-paid system with no contracts, no monthly fees and low VoIP rates, you will have the comfort and ease you need.
Signup with a VoIP provider for call centers if you are using predictive dialers such as Vicidial or Goautodial for your telemarketing campaigns.
Advanced VoIP / PSTN Services • Lowest competitive rates based on your call volume • Best technical support • Access your own CDR (Call Detail Records) online • Top of the line quality of service • Tech support configuration help
Switch2Voip offers high quality, low cost, excellent voice and fax services to their customers in every country!
At Switch2Voip we take pride in offering Wholesale and Retail customers the advantage of a global network without the hassle of building, managing and directing it themselves. Service Providers also gain and receive the advantage of the direct interconnections with world’s leading telecommunication carriers.
We can connect your calls using VoIP (H.323 and SIP protocols) coming from any country on the planet through Internet. Continue reading
Business VoIP service provider: Switch2Voip provides VoIP service for businesses looking to make calls all over the world at very low rates. http://www.switch2voip.us/blog/21-voip-for-business-ippbx
Test a SIP Account that offers better price, reliability and quality when you compare it to local telephone companies and their Business VoIP Solutions.
Start making calls with just $5 and enjoy prepaid phone service for personal use, business or call centers on a pay-as-you-go basis.
Read more information on our previous article: Best VoIP provider for business
According to Wikipedia:
An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.
ITSPs provide services to end-users directly or as whole-sale suppliers to other ITSPs.
https://en.wikipedia.org/wiki/Internet_telephony_service_provider
A digital virtual voice solution for Business and CallCenters that gives the user the option of calling their clients at a portion of the cost. All VoIP rates are under 1 cent per minute.
Increase your work rate and capability, your business will become more efficient and productive using a hosted Virtual Call Center at only $15 per extension/month per agent.
Switch2Voip allows you to keep the current software and technology you’re using in your business, with just a few configuration changes in your Auto-Dialer or IP-PBX and you can start making calls at very low cost.
With maintaining your same Business phone number and technology and being able to lower costs, your company will be able to communicate more efficiently and effectively with its clients.
From your home or office, using your internet connection you will be able to modify your account settings and features fast and easy by just logging into your account without any limitations.
Switch2Voip has a reputation and success rate due to its business friendly characteristics, competitive rates, simplified billing and customer service.
Switch2Voip Virtual Call Center solutions provides your VoIP Call Center with the same features without the maintenance cost, the perfect solution.
A Virtual Call Center increases the production of your business while reducing the IT staff and your start up costs.
Having a Virtual Call Center Hosted Auto-Dialer is a more viable option over having an In-House server. With our hosted solution 99% uptime is guaranteed.
24/7 Tech Support included even if you only rent 1 Seat from us.
You do not have to pay hundreds of dollars on IT’s installing and configuring an Auto-Dialer on your call center, if you only need to run a telemarketing campaign with a few sales agents, now you have the option to rent the extensions you need and whenever you need them and your auto-dialer will be ready in the fly.
The rented Virtual Call Center VICIdial solution is best for fast start-ups with instant advantages. You will save on:
With the help of our rented Hosted Virtual Call Center you will receive 99.99% system uptime and fault tolerance.
The server we will setup for your Virtual Call Center will be on a network that is state of the art with Fiber backbone capable of handling any volume of calls, UPS backups, RAID for fault tolerance and 24 hour proactive monitoring.
Voice blast or voice broadcast is a word that refers to the concept of sending to many different people the same recorded voice message.
This is a similar concept as a “robocall” which most of the people are used to.
Whether any political group wants to send out a recorded message, survey or call or any school or local organization wants to deliver a message to their employees or parents, you can use our Virtual Call Center for your marketing needs.
| AutoDialer Features | |
|---|---|
|
Auto Dialing
|
 |
| Leads Uploading | |
|
Agent Monitoring
|
|
| Voice Recording | |
| Scalable Seats | |
| Full Tech Support | |
|
Storage
|
60 days
|
|
Setup Fee
|
$50.00
|
|
Monthly Fee
|
$15 per seat
|
| Click “Join Now” to Add/Delete seats to select how many extensions you want before completing your order in the shopping cart. | |
| PBX in the Cloud | |
|---|---|
|
Custom Greeting
|
|
| Agent Extensions | |
|
Call Recording
|
|
| Call Forwarding | |
| Scalable Extensions | |
| Full Tech Support | |
|
Storage
|
60 days
|
|
Setup Fee
|
$50.00
|
|
Monthly Fee
|
$15 per seat
|
| Click “Join Now” to Add/Delete seats to select how many extensions you want before completing your order in the shopping cart. | |
http://ow.ly/4n73Z9 $15 per seat, you can start from 1 seat. A Hosted Predictive Dialer is the perfect solution! It gives you the same features and convenience without ownership cost and maintenance!
Virtual Call Center boosts and increases the production of Cloud Call Centers while lowering start-up costs.
Olympia, WA — (SBWIRE) — 01/30/2014 — Switch2Voip provides VoIP for Call Centers in Philippines. http://ow.ly/38hMiC
Switch2Voip now offers VoIP in Philippines. Call centers in Philippines can now benefit with the Switch2VoIP.us services as it can save a substantial amount of money when as compared to the cost incurred for long distance calls made via conventional methods. Switch2VoIP.us offers highly reliable and affordable VoIP services at competitive rates without compromising on quality.
Switch2VoIP guarantees its clients in the Philippines a well trained support team that is accessible 24/7. Call centres in the Philippines can switch to the advanced and innovative service offerings of Switch2VoIP that has a global presence and vast experience in terms of rendering quality service in the field of VoIP and to call centres in specific.
Call Centres and contact centres in the Philippines now have access to top levels of uptime and support from a leading VoIP service provider now at their door step. Switch2VoIP is compatible with all Virtual Call Center Asterisk based platforms such as Vicidial, Goautodial, ViciBox, VicidialNow, Elastix, etc.
VoIP in Philippines through Switch2VoIP now makes operating call centre call volumes, managing the number of channels as well as SIP trunking account so as to make VoIP calls. Switch2VoIP is certainly the VoIP service provider that can understand One’s needs or provide the number of channels one need for the number of calls per second that call centres typically make.
Switch2Voip integrates into the VoIP Philippines market with a solution for Call Centers that allow competing and reducing operational costs while providing top quality VoIP services and has the unique ability to make unlimited concurrent calls from their auto-dialers without the need of opening more than 1 account.
Switch2VoIP offers other valuable services like ‘VoIP Pay As You Go’ billing model that provides the best quality of service without commitments or agreements. Also the premium A-Z VoIP termination can be requested for. Wholesale VoIP rates without having to buy a big amount. Other extended services besides VoIP phone service include bulk SMS messaging, VoIP phone numbers, international DID’s and so on. Further the call centres can add balance online with easy payment options.
Switch2Voip offers VoIP in Philippines some of the top experts in the VoIP industry and network engineering so call centers and business customers have the advantage of having their problems relating to communications are handled by efficient and knowledgeable team of experts.
Track Records. Switch2Voip proudly says that they have been offering SIP trunking for call centers since 2006 and that their customers are spread in as many as in 55 countries. Further, their clients include not only call centers but large business and residential VoIP customers as well. They further assert that their VoIP solutions help in managing billions of calls every year and these solutions come with 99.999% accuracy.
About Switch2Voip. Switch2Voip team of experienced professionals is committed to providing high quality and reliable VoIP services to customers worldwide. This service provider has been successfully delivering wholesale VoIP solutions since 2007 and has a global presence is now makings its presence.
https://www.switch2voip.us/blog/180-philippines-call-center-voip
https://www.switch2voip.us/blog/160-voip-for-call-centers-in-philippines-the-benefits
Best VoIP Providers Information for Business 