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asterisk sip trunk provider
Some of our key SIP Trunking features:-
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✓ Scale as per your need
✓ Flexible Pricing options to choose
✓ Reliable service & support
✓ Easy account management
✓ Multiple compatible platforms
✓ Largest Phone Number inventory
✓ SMS/MMS enabled Phone Numbers
✓ Number Portability
✓ Easy Integration
✓ 30-day money back guarantee
Asterisk Most frequently used commands
Here are some of the most commonly used Asterisk Commands:-
asterisk –rvvvv : Enter Asterisk cli
sip show peers : Check registered sip users in asterisk
sip set debug on : Enable sip debugging
sip set debug ip x.x.x.x : Enable sip debug for IP x.x.x.x
sip set debug peer xxxx : Enable sip debug for extension xxxx
sip set debug off : Disable sip debug
core stop now : stop asterisk service from cli
core restart now : restart asterisk service from cli
core show version: Check version of asterisk
sip show channels : check running sip channels
core set debug 5 : set the core debug to level 5
core set verbose 9 : set verbosity level to 9
reload : Reloading complete asterisk configuration
dialplan reload : reload dialplan only
sip reload : reload sip settings only
dialplan show : shows all the dialplans in the system
core show applications : list all the available dialplan applications in asterisk
core show functions : list all the available dialplan functions in asterisk
asterisk –rx “command” : Running asterisk commands outside of CLI
asterisk -rx “sip show channels” : Will Display running channels.
Why to manage a phone system
when you can get for free.
Are you having an audio issues in your Asterisk?
Well it’s a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks
Learn how you can limit number of simultaneous calls on Asterisk based SIP Trunk. If you want to limit the number calls for your SIP peer or friend in Asterisk use call-limit in your trunk configuration.read more
This is a very common requirement that route the calls to Voice-mail after office hours. Or you can transfer the calls to your cell phone after certain time say 6:00pm. In Asterisk you can also control the call location based on time and date.read more
Visit SIP Trunking Pricing to see which plan best suits your business!
With so many options to pick from it can often be hard to decide what’s best.
Our plans have been packaged together to give you optimum output.
Our SIP Trunks are Compatible with wide range of PBX & Platforms.