[Asterisk-Users] iax invitation problem

[jonny hashem]

i have opened an account with  callshopcompany,and
when ive tried to send calls by the sip i had a
message show an asterisk invitation problem  i had
these sip configuration:
                       sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=XXXXX
secret=XXXXXXX
 
Then i tried to add these lines and it worked :
                   sip.conf

[callshop]                
type=peer
host=sip.callshopcompany.com
username=XXXXXX
secret=XXXXXX
fromuser=XXXXXX
usereqphone=yes
canreinvite=no
nat=yes
insecure=invite
insecure=port
port=5060

but when i tried to send calls using iax the call
authenticated and then failed giving this:

-- Executing Dial("OSS/dsp",
"IAX2/call/0017046872001") in new stack
    -- Called call/0017046872001
    -- Call accepted by 213.61.187.150 (format g729)
    -- Format for call is g729
    -- Hungup 'IAX2/call/2'
  == No one is available to answer at this time
Oct  2 23:32:37 WARNING[18702]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
 << Hangup on console >>

The iax.conf file is like this:

[call]
type=peer
username=875630553
secret=darWish472
host=sip.callshopcompany.com





		
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