asterisk users - Feb 2006 - Asttapi - what's wrong?

When I try to call from asttapi one number, I get message "No one is
available to answer at this time (1:0/0/0)". Immediately after that I try
to call the same number from SIP phone (the same one that is used with asttapi)
and call goes true.

What have I done wrong?

This is how it looks on CLI.

****

  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'tomo' logged on from 10.0.0.203
       > Channel SIP/341-062e was answered.
    -- Executing Dial("SIP/341-062e",
"OOH323/0989970434@xxx.xxx.xxx.xxx") in new s
tack
    -- Called 0989970434@xxx.xxx.xxx.xxx
  == No one is available to answer at this time (1:0/0/0)
    -- Executing Hangup("SIP/341-062e", "") in new stack
  == Spawn extension (sip, 00989970434, 2) exited non-zero on
'SIP/341-062e'
  == Manager 'tomo' logged off from 10.0.0.203
    -- Executing Dial("SIP/341-9e85",
"OOH323/0989970434@xxx.xxx.xxx.xxx") in new s
tack
    -- Called 0989970434@xxx.xxx.xxx.xxx
    -- OOH323/85.114.35.42-b1b4 is ringing
  == Spawn extension (sip, 00989970434, 1) exited non-zero on
'SIP/341-9e85'

In article <79337FA3BA80774F9FE87BF809C9F3562CA78B@stipe.lama.hr>,
tparcina@lama.hr says...
> When I try to call from asttapi one number, I get message "No one is
available to answer at this time (1:0/0/0)". Immediately after that I try
to call the same number from SIP phone (the same one that is used with asttapi)
and call goes true.
> 
> What have I done wrong?
Solved!

Problem vas that manager adds default caller ID (not the one that was defined in
sip.conf for the phone from which I'll will speak). And I need to sent to
provider specific caller ID.

Now, I have question. In agents conf, can I define Caller ID for every user
(manager)? If not, that is something that defiantly should be implemented.


--
Tomislav Parcina
tparcina#lama.hr