Thursday, July 29, 2010
New Call Rates - August 2010
We are pleased to introduce our new call rates plan for August 2010. Many destinations are now cheaper than ever to call. Please check our Call Rates page for charges to all destinations.
Friday, July 16, 2010
Mydivert.com on NetVibes
Netvibes is a multi-lingual Ajax-based personalized start page, or personal web portal, now has its very own mydivert dashboard, thanks to our work experience team member, Stephen Bray.
check it out at www.netvibes.com/mydivert
Well done Stephen... now put the kettle on make the coffee!
check it out at www.netvibes.com/mydivert
Well done Stephen... now put the kettle on make the coffee!
Free Polish VoIP Numbers and Calling Credit
Special Offer - Free Polish VoIP Telephone Numbers and International Calling Credit
Stay close to your family, friends, and business contacts by having a local Polish Telephone Number that forwards calls wherever you want, to your office, home, or mobile, anywhere in the world.
With a virtual Polish number on your phone or mobile your callers pay Local Polish Call Rates when they call you.
In May 2010 our call forwarding service partnered with the largest VoIP provider in Poland to bring a geographical range of Polish VoIP telephone numbers to market via our website www.mydivert.com. Anyone familiar with VoIP telephony, and incoming phone numbers, will know how these can be used to be contactable on a local Polish phone number regardless of where the person is actually located. If not, please see our website for details.
Free Trial Account – 1 Local Polish Telephone Number + 50 cent Calling Credit
We are pleased to introduce, for a limited period*, an offer of Free Trial Accounts for our Polish Virtual Numbers and Call Forwarding Service. Interested parties are invited to send their name & email address to us at polishsales@mydivert.com. We will then create an account for you at www.mydivert.com and send the user details to the email address supplied.
Once the user logs in, they are able to request a Polish telephone number from any area code. We will add the number to the account Free of Charge with a 1 month subscription and also add 50 cent calling credit (required for forwarding incoming calls, or outbound calling).
We hope that users will enjoy the service and decide to remain after the first month trial.
Virtual Polish telephone numbers purchased at MyDivert.com can also be used with your own SIP provider or PBX.
* Offer of Free Trial Accounts is limited to the first 300 applicants
Stay close to your family, friends, and business contacts by having a local Polish Telephone Number that forwards calls wherever you want, to your office, home, or mobile, anywhere in the world.
With a virtual Polish number on your phone or mobile your callers pay Local Polish Call Rates when they call you.
In May 2010 our call forwarding service partnered with the largest VoIP provider in Poland to bring a geographical range of Polish VoIP telephone numbers to market via our website www.mydivert.com. Anyone familiar with VoIP telephony, and incoming phone numbers, will know how these can be used to be contactable on a local Polish phone number regardless of where the person is actually located. If not, please see our website for details.
Free Trial Account – 1 Local Polish Telephone Number + 50 cent Calling Credit
We are pleased to introduce, for a limited period*, an offer of Free Trial Accounts for our Polish Virtual Numbers and Call Forwarding Service. Interested parties are invited to send their name & email address to us at polishsales@mydivert.com. We will then create an account for you at www.mydivert.com and send the user details to the email address supplied.
Once the user logs in, they are able to request a Polish telephone number from any area code. We will add the number to the account Free of Charge with a 1 month subscription and also add 50 cent calling credit (required for forwarding incoming calls, or outbound calling).
We hope that users will enjoy the service and decide to remain after the first month trial.
Virtual Polish telephone numbers purchased at MyDivert.com can also be used with your own SIP provider or PBX.
* Offer of Free Trial Accounts is limited to the first 300 applicants
Sunday, July 4, 2010
Irish Virtual Numbers
We are pleased to introduce an expanded range of geographical coverage for Irish VoIP Virtual Numbers.
Now you can have an incoming VoIP number for your virtual office in Dublin for example, or for your friends and family back home to be able to call you at local rates from their district.
The Irish virtual numbers are charged at a flat rate €3.50 per month (incusive sales tax). This allows you to receive unlimited incoming VoIP calls. You can also take advantage of our outbound call routes if you wish to place calls at very cheap rates. Caller ID number on outbound calls is supported.
Calls to your Irish virtual number can be forwarded to your mobile, landline telephone, VoIP telephone, SIP device, or PBX such as Asterisk or FreePBX.
Now you can have an incoming VoIP number for your virtual office in Dublin for example, or for your friends and family back home to be able to call you at local rates from their district.
The Irish virtual numbers are charged at a flat rate €3.50 per month (incusive sales tax). This allows you to receive unlimited incoming VoIP calls. You can also take advantage of our outbound call routes if you wish to place calls at very cheap rates. Caller ID number on outbound calls is supported.
Calls to your Irish virtual number can be forwarded to your mobile, landline telephone, VoIP telephone, SIP device, or PBX such as Asterisk or FreePBX.
Saturday, July 3, 2010
Jordan Amman Virtual Numbers
We are pleased to reintroduce the Jordan Amman Virtual Number range to our DID coverage.
The Jordanian DID had previously suffered from poor international interconnect that made them only reachable from within Jordan. These interconnect issues have now been resolved by the local carrier and we are now supplying these numbers once again.
Jordanian virtual numbers are not permitted for use with calling card systems.
The Jordanian DID had previously suffered from poor international interconnect that made them only reachable from within Jordan. These interconnect issues have now been resolved by the local carrier and we are now supplying these numbers once again.
Jordanian virtual numbers are not permitted for use with calling card systems.
Monday, June 28, 2010
Debugging Asterisk Tips
It is a common problem that people starting out with Asterisk PBX find it difficult to diagnose where problems arise. Here is a quick blog that may help someone in that situation.
To debug asterisk you need to get hold of the asterisk messages and any other log info you can. This isn't always so simple if you have a busy system as so many records are created.
If you have a lot of calls its important to be very quick when you place a test call and capture output so you won't have multiple calls in the same capture.
Log into your asterisk server via ssh.
Usually I then type :
asterisk -rx "sip set debug on"
asterisk -rcvvvvvvvvvvvvvvvvvvvvv> /usr/local/admin/htdocs/debug.txt
(note 1: you can use asterisk -rx "sip set debug on IP" if you know a particular IP, or use PEER. Type help from the command line for options)
(note 2: replace /usr/local/admin/htdocs/debug.txt with some path on your server. If the path is to htdocs then you can easily download the capture using a browser)
And after that I call the number that I want to test and then I hangup when the call goes through. Then I quickly stop the asterisk capture with CTRL+C.
After that I verify the Asterisk capture. I just download it and view in a text editor.
In this situation I only verified the Asterisk log but in another situation I might have to check the SIP packets. To do that I use tcdump.
If tcpdump is not installed you can run 'yum install tcpdump' on a linux system to install.
I then run something like
'tcpdump -p udp -s 4096 -w /usr/local/admin/htdocs/capture.pcap'
to capture all traffic.
or, 'tcpdump -w /usr/local/admin/htdocs/capture.pcap -s 0 host sip.avoipcarrier.net' to get a dump for just one host.
After I place a call from an extension to an specific number I stop the capture with CTRL+C and I download the capture to my computer.
Then using a tool like Wireshark (you can find that on the internet since its a free tool) I verify the call flows and view the SIP messages.
I guess this is a basic howto on how to debug basic things. if you want to learn more check some articles regarding basic call flows and how SIP handles basic transactions.
Also check some tutorials about Wireshark (source/destination port/address, how to debug rtp, sip messages, filters, jitter...).
I hope this helps.
To debug asterisk you need to get hold of the asterisk messages and any other log info you can. This isn't always so simple if you have a busy system as so many records are created.
If you have a lot of calls its important to be very quick when you place a test call and capture output so you won't have multiple calls in the same capture.
Log into your asterisk server via ssh.
Usually I then type :
asterisk -rx "sip set debug on"
asterisk -rcvvvvvvvvvvvvvvvvvvvvv> /usr/local/admin/htdocs/debug.txt
(note 1: you can use asterisk -rx "sip set debug on IP" if you know a particular IP, or use PEER. Type help from the command line for options)
(note 2: replace /usr/local/admin/htdocs/debug.txt with some path on your server. If the path is to htdocs then you can easily download the capture using a browser)
And after that I call the number that I want to test and then I hangup when the call goes through. Then I quickly stop the asterisk capture with CTRL+C.
After that I verify the Asterisk capture. I just download it and view in a text editor.
In this situation I only verified the Asterisk log but in another situation I might have to check the SIP packets. To do that I use tcdump.
If tcpdump is not installed you can run 'yum install tcpdump' on a linux system to install.
I then run something like
'tcpdump -p udp -s 4096 -w /usr/local/admin/htdocs/capture.pcap'
to capture all traffic.
or, 'tcpdump -w /usr/local/admin/htdocs/capture.pcap -s 0 host sip.avoipcarrier.net' to get a dump for just one host.
After I place a call from an extension to an specific number I stop the capture with CTRL+C and I download the capture to my computer.
Then using a tool like Wireshark (you can find that on the internet since its a free tool) I verify the call flows and view the SIP messages.
I guess this is a basic howto on how to debug basic things. if you want to learn more check some articles regarding basic call flows and how SIP handles basic transactions.
Also check some tutorials about Wireshark (source/destination port/address, how to debug rtp, sip messages, filters, jitter...).
I hope this helps.
Thursday, June 24, 2010
Using 3CXphone
3CXPhone is a free VoIP phone that allows you to use your PC or laptop as a phone.
You can connect 3CXPhone to a VOIP provider such as mydivert.com to make calls to any VoIP, mobile or landline number. By adding an incoming virtual number to your mydivert.com account you can be instantly contactable on a local number from over 65 countries worlwide.
To download 3cxphone simply visit the 3cx website. The download is 3MB and installation only takes a couple of minutes.
The 3CXphone compares very favourably to other softphones. Time to establish SIP registartion is fast and dialed calls are established almost immediatly. The phone interface is attractive and user friendly. All the usual features are provided such as voicemail alerts, address book, call records etc.
To configure 3CXphone with mydivert.com you should create a new SIP profile and add your account details as below:
The advanced settings can then be set. We added *95 as the voicemail access number, and checked that the SIP transport method is set to UDP (important!). As we are running the sofphone nehind a NAT router we set a short registration period (2 - 5 mins) to help with keep-alives and maintain the phoen presence.
As we want the phone to send DTMF tones that we will use when logging into out voicemail system we unchecked the INBAND and SIPINFO DTMF signaling. These methods are not supported.
That's all. The softphone works very well and we would rate it with 5 stars.
3CX also offer complete pbx systems for windows which work well with our sip trunking service.
You can connect 3CXPhone to a VOIP provider such as mydivert.com to make calls to any VoIP, mobile or landline number. By adding an incoming virtual number to your mydivert.com account you can be instantly contactable on a local number from over 65 countries worlwide.
To download 3cxphone simply visit the 3cx website. The download is 3MB and installation only takes a couple of minutes.
The 3CXphone compares very favourably to other softphones. Time to establish SIP registartion is fast and dialed calls are established almost immediatly. The phone interface is attractive and user friendly. All the usual features are provided such as voicemail alerts, address book, call records etc.
To configure 3CXphone with mydivert.com you should create a new SIP profile and add your account details as below:
The advanced settings can then be set. We added *95 as the voicemail access number, and checked that the SIP transport method is set to UDP (important!). As we are running the sofphone nehind a NAT router we set a short registration period (2 - 5 mins) to help with keep-alives and maintain the phoen presence.
As we want the phone to send DTMF tones that we will use when logging into out voicemail system we unchecked the INBAND and SIPINFO DTMF signaling. These methods are not supported.
That's all. The softphone works very well and we would rate it with 5 stars.
3CX also offer complete pbx systems for windows which work well with our sip trunking service.
Subscribe to:
Posts (Atom)
