ok i feel stupid but.....

[vmhost303]

OK,

I got my server online but, i talked to someone at vicidial about this. I forget who i talked to but they had stated i wouldn't need FREEPBX to also run soft phones. I get connected to the asterisk server under a sip number i created to test out the system. I have tried numerous dial plans ones even provided by the trunk provider which is flowroute. My dial plan looks like this (minus my username and passwords:

Code: Select all


"ACCOUNT ENTRY"
type=friend
secret="PASS FROM FLOWROUTE"
username="USERNAME FROM FLOWROUTE"
host="flowroutes site"
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain="flowroutes Site"

"Registration String"
register => "USER":"PASS"<at>Flowroutes Site

"Globals String"
TESTSIPTRUNK = SIP/testcarrier


Now on the dial plan i have used everything from flowroute and from the vicidial manager book.

Now it boils down why can't i make a simple direct line call from the logged in sip on eyebeam to my cell phone?

Also in the code all site info was changed haven't been a member long enough to be able to post them..............

Ok so i am editing this because i am watching the terminal as i try to place a call i use the "9" for a outside line and i get more of a responce. It is saying "DEADAGI" and killing the call. Any idea's?

I have also done a "sip show registry" and the asterisk server is registered with flowroute and is up and running but i canot get it to place a outside direct call.

Ok i also tried dialing numbers from a list and got the same thing "CALL REJECTED"

[mflorell]

admin.php version and build?

Asterisk CLI output of you making the call?

[vmhost303]

The versions are all of the version's included with redux 1.8. this server has yet to actually make its first dial call. as for the info given from the asterisk termin it reads:

Code: Select all

AGI SCRIPT agi://127.0.0.1:4577/call_log--HVcauses--PRI----MODEBUG----20-----CHANUNAVAIL--------completed, returning 0


also i am finding with it a [WARNING 31867]
and in just watching it i find a
[31913]: utils.c:966 ast_carefulewrite: write() returned error
then 2 errors broken pipe and connection reset by peer

The whole server setup is exactly how the redux 1.8 install said to do it

[mflorell]

That's not enough Asterisk CLI output.

[vmhost303]

That's not enough Asterisk CLI output.

***CAN'T POST PER THE SYSTEM OF NOT BEEN HERE LONG ENOUGH****

There is a screen shot from the terminal output. This is what it out puts as i try to just dial a outside line. I am only connecting to the sip and trying to dial out. I am using 9 as the trunk extension. I tried every number for the trunk extension but this is the only one that doesn't call it a extension on the system.


PM me and i will send you the link if you can help.

[mflorell]

"Unable to create channel type SIP"

Not sure if I've seen that before exactly, since chan_sip should be loaded by default in Asterisk.