I hope I can help. I think you will need to get log files before anyone can help you.
In order for you to get enough info on the log file, do the following.
1. get into astersik command line. type asterisk -vvvr
2. set verbose level to 10. type core set verbose 10
3. set debug level to 10. type core set debug 10
4. you may need to reload the asterisk. Do not exit. Just type reload
5. If you are using sip termination, you may want to have some debuging info on the sip. type sip debug peer (the sip name of your sip provider).
6. Make an outbound call.
7. Look/read through /var/log/asterisk/full. It is going to be overwhelming beacuse there will be a lot of data. However, with enough patient, you may spot your error.
definitely requires close inspection of logs - preferably by someone familiar with VoIP 'mechanics' so to speak - since there can often be messages (or more likely, a group of them) that appear to be either informational or duplicates - and can actually be errors or unexpected events.
likely things happening though are
a) dial-plan not matching correctly for outbound calls on the trunk
b) caller-id being set/overridden to something not supported by your VoIP Trunk
c) Codec Negotiation error/mismatch
d) NAT issues
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I know this is going to sound stupid, but did you set a proper outbound route so your phones route to a specific outbound trunk?
Is the outbound trunk properly registering with your VOIP Provider?
█ Matthew Rosenblatt, and I do lots of things.
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