Hi everyone, Im new to VOIP. Right now I’m using freepbx 2.6.0.RC2.1 with asterisk 1.4.26.2 . I’m having difficulties understanding inbound route. Example, Ive set my extension 6000 to have a destination which is extension 6000. However, when I set it to have destination to my IVR, nothing changes and it will still revert back to my extension being unavailable etc. I hope someone can enlighten me on my mistake here, much appreciated and thanks.
Did you ever get an answer? Your question is not clear. You say you set extension 6000 to have a destination of 6000. I don’t think that is what you mean. You start with a trunk and that trunk has a destination which may be an extension or IVR among several other options. That destination is set in inbound routes. If you are still looking, be a little more specific with your question.
Thanks for your reply,
Thanks for your slight explanation, can I safely say that inbound routes are the path destination for calls that are received from outside the trunk? I think my question is on internal calls as my trunk is not configured to talk to another machine yet.
Inbound routes are tools to control the destination of calls through the trunk. Think of the trunk as a pipe. The call enters the Asterisk system through the trunk. Then the inbound routes are smaller pipes (perhaps one to the kitchen, one to the bathroom another to the washer). An inbound route is not a destination, it is a means to direct the call to the desired destination. If you only want to use Asterisk for internal calls, you wouldn’t use a trunk or destination either. Internal calls are like intercoms. The calls go direct from extension to extension. So if you only have 6000, you need to create more extensions, which means more phones. In the simplest sense, a trunk is an incoming line. This probably makes it foggier, but if you will describe what you are trying to do, me or someone else can help more. For example, do you plan to be able to accept calls from the public telephone network, if not, you probably don’t need Asterisk. What kind of trunk will you have. Will it be regular telephone line from local phone company or SIP or other. If you get a SIP trunk, asterisk can handle all outgoing and incoming calls. If you want to use a public phone line, you will need some hardware to enable that. You mention using an IVR. That would normally be used for incoming calls. Also, what kind of phones are you using? There is some good documentation on the FreePBX site that you need to study. Installation and configuration is very doable, but also very complicated. Its not something you can learn in a few days.
Thanks for your reply,
I am using Asterisk to make calls between 2 machines. However I have been very unsuccessful with my trunk configuration. I am only able to use softphone such as xlite to make calls.
Machine 2 Staff
Trunk Name: staff2student
Peer Details:
context=from-internal
host=xxx.xxx.xxx.xx
qualify=yes
secret=<removed by forum admin>
type=peer
username=staffuser
User Context:studentuser
User Details:
context=from-internal
host=xxx.xxx.xxx.xx
secret=<removed by forum admin>
type=user
Machine 1 Student
Trunk Name: student2staff
Peer Details:
context=from-internal
host=yyy.yyy.yyy.yy
qualify=yes
secret=<removed by forum admin>
type=peer
username=studentuser
User Context: staffuser
User Details:
context=from-internal
host=yyy.yyy.yyy.yy
secret=<removed by forum admin>
type=user
The above are the codes that I am trying to use for both Staff and Student Machines respectively, your comments on how I might be able to improve from my mistakes is greatly appreciated.
My scope involves making the trunk work between these 2 machines with basic extensions, IVR to receive calls, making system recordings, day/night control with time condition and above all these, click to dial which I am trying to use with asterCRM which isnt going smoothly either.
Thanks for your help, I have a better understanding of how the trunk and inbound routes work.
However I’m a little confused about how the FreePBX works. Does this mean that internal calls are not affected by the FreePBX trunk and route configurations?
But first, if the info you posted is the actual, change your passwords IMMEDIATELY. I’m not knowledgeable about calls from server to server but will think about it. JUST BE SURE TO CHANGE PASSWORDS
These webpages use the PBX in a Flash distribution which uses FreePBX as the management tool - same process. Here’s How to set up an IP Phone, here’s how to setup X-lite and finally how to get the IVR working
Thank you all for your help, I am confident that my IVR is working normally with the assigned feature code. However is there a way whereby when I make a call to for example extension 6000, the IVR takes over automatically?
Regarding my trunk configuration, I have tried many different variations but it still gives me “your call cannot be proceeded as dialled”
Machine A
Outgoing Settings
Trunk Name: systema
PEER Details:
context=from-internal
host=XXX.XXX.XXX.65
qualify=yes
secret=XXXXXXa-pass
type=peer
username=systema-user
Incoming Settings
USER Context: systemb-user
USER Details:
context=from-internal
host=XXX.XXX.XXX.65
secret=XXXXXXb-pass
type=user
Machine B
Trunk Name: systemb
PEER Details:
context=from-internal
host=XXX.XXX.XXX.77
qualify=yes
secret=XXXXXXb-pass
type=peer
username=systemb-user
User Context: systema-user
User Details:
context=from-internal
host=XXX.XXX.XXX.77
secret=XXXXXXa-pass
type=user
I did some editing to my eariler configuration for both trunks, however it still isnt working. I’m hoping for anyone to be able to spot my mistake that I am unable to do so.
outbound route for systema
dial pattern 80XX <----systema extension pattern
trunk Sequence IAX2/systema
outbound route for systemb
dial pattern 60XX <----systemb extension pattern
trunk Sequence IAX2/systemb
Above are my outbound routes. Any help is appreciated.