I have a call center, and I’m using the asterisk as my default system, I want to assign a phone number to the asterisk system to receive external inbound calls through it.
Contact a traditional telephone operator and rent an ISDN T1 or E1 connection from them, and install an ISDN FXO card.
Contact a VoIP provider and rent an incoming telephone number from them and connect to them via an IP service provider.
Buy a GSM gateway, and a SIM and subscribe to a mobile network number (some countries may impose restrictions on this to prevent mobile network capacity being wasted on this.
Buy a GSM dongle,then as above.
Most small users go for the voice over IP option.
You can also rent an analogue line, and use it with an analogue FXO card, or an analogue line to SIP gateway device, but the signalling on analogue lines is not really suitable if you have any automated calls.
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Thank you so much for your response, I forgot to mention that what I want exactly is to receive these phone calls to my softphone on my computer, so what I need is the right configurations for inbound calls made from external users and I want those calls to go through a Dialplan so I could add IVR or Music on hold to the call…etc
That is very basic use, so you should be looking at the example configurations, once you have chosen your service provider. You could also look at https://www.asteriskdocs.org
Whilst that web site can be a little dated, this requirement is so standard that nothing significant should have changed.
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Thank you again for your reply, the links above does not work for me ): But Hope you can help me a little bit.
Yes i have Sip accounts and i know how to add them to the sip.conf, and my provider gave me a phone number with credintals as showen bellow. I need to know how to configure the sip.conf and extensions.conf for inbound calls coming from clinets to our softphone (Agent1).
i will give you a scenario to help guide me through the configurations:
lets say this is my phone number provided to me :
Phone number: 49800000000
user: agen100000
pass: qwert123
sip.conf
[Agent1]
type = friend
host = dynamic
context = inbound-calls
username = myusername
password =mypassword
qualify = yes
nat = force_rport,comedia
extensions.conf
[inbound-calls]
exten => _4980000000X!,1,NoOp(Call For agent1)
exten => _X.,n,Dial(SIP/Agent1)
exten => _X.,n,Hangup()
Use it with http:, rather than https:
What is going wrong with the above? Please provide logs of the inbound calls with at least verbosity 3, so we can see how far the call gets in the dialplan, and why it stops.
Why is the agent in context inbound_calls? Normally you would want local users to be able to make chargeable calls, but you wouldn’t want incoming callers to do so, so you would not normally give that capability in the inbound calls context.
Please explain your network configuration, and therefore why you need to override the default NAT handling of auto_force_rport,auto_comedia.
I’m confused about the user name for what I presume to be your ITSP account. I’d expect the phone number, an arbitrary number, or something derived from your business name. In any case, you haven’t provided your sip.conf section for the provider.
Why are you not using chan_pjsip?
PS Please mark up logs and configuration, for the forum, as pre-formatted text, e.g. using </>.
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