Softphone SIP to SIP unavailable number

Ubuntu 8.04
FreePBX 1.5.1

I followed the guide and it all looks ok.

I’m having problems ring from softphone to softphone. One is X Lite on Windows and the other is Ekiga on Ubuntu.

I created 2 extensions for each softphone using all default values except the secret and User Extension fields.

Do I need to do anything else? The 2 phones resister with no problem and I get the PBX voice too. When I try ringing from one softphone to the other I get the message “The person you are calling is unavailable. Please try again” and says the call failed. The problem exists dialling from either of the phones to each other (not at the same time).

Am I missing something? I have not set other settings in FreePBX.

Please re-verify the details as there was never version 1.5.1 of FreePBX. What version of asterisk are you using? I’m guessing you meant 2.5.1, but making assumptions on anothers posting is dangerous these days so please clarify.

Also posting a call trace from the cli with the verbose set to at least 3 would help.

How did you verify that the extensions are registered properly?

Sorry it is 2.5.1.

I started asterisk using the command:
sudo /usr/sbin/asterisk -vvvvvv
to give me verbose and see what happens when I call softphone to softphone but it worked! However when if I start Asterisk as a daemon I get the same problem as before. Is this something to do with permissions?

Are there log files I can use when I start Asterisk as a daemon so I can post on here?

I verfied the extensions were registered properly by calling the voice mail (after going back and turning it on) and I could type my pin and login to the voice mail from both softphones.

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Thanks and regards
SoftPhone User

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