And what should i configure them to? I tried setting my SIP/peer for my outgoing lines, but it doesn't seam to do anything. What' is Trunk and what should it do? Any example configurations?
The trunk button is like an extension button but for trunks
It is supposed to show activity about your trunks.
If the channel name is correctly identified, and you have active calls for that channel, you will see the button in orange and a text like "1 channel in use", or "10 channels in use" if you have 10 ongoing calls over that trunk.
A trunk is a good place to do channel overloading, where you define multiple channels for just one button, example:
[DAHDI/1]
type=trunk
label=4 Analog lines
channel=DAHDI/2
channel=DAHDI/3
channel=DAHDI/4
Any call that is active on any of the channels listed will count for that trunk button.
Regarding your problem, perhaps you are not using the proper channel name.
There is a common issue with sip trunks and asterisk: 80% of installs have a peer defined for outbound, while inbound calls are not matched and coming into asterisk as anonymous or guest calls. In that case it will be really hard to track those calls as asterisk will name the channels something like SIP/ip.add.re.ss instead of SIP/yourProvider
Yes. That's it. I've always had trouble with incoming calls. Whatever I've done they've always been coming in on the default context. Guess I need to read some docs. Thanks.
Comments
It is supposed to show activity about your trunks.
If the channel name is correctly identified, and you have active calls for that channel, you will see the button in orange and a text like "1 channel in use", or "10 channels in use" if you have 10 ongoing calls over that trunk.
A trunk is a good place to do channel overloading, where you define multiple channels for just one button, example:
[DAHDI/1]
type=trunk
label=4 Analog lines
channel=DAHDI/2
channel=DAHDI/3
channel=DAHDI/4
Any call that is active on any of the channels listed will count for that trunk button.
Regarding your problem, perhaps you are not using the proper channel name.
There is a common issue with sip trunks and asterisk: 80% of installs have a peer defined for outbound, while inbound calls are not matched and coming into asterisk as anonymous or guest calls. In that case it will be really hard to track those calls as asterisk will name the channels something like SIP/ip.add.re.ss instead of SIP/yourProvider
Best regards,