[Asterisk-Users] iax invtation problem
jonny hashem
jonnyhashem at yahoo.com
Sun Oct 16 10:43:39 MST 2005
i had a sip invitation problem with my voip provider
and here the message that was shown :
Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890
handle_response: Forbidden - wrong password on
authentication for INVITE to '"asterisk"
<sip:XXXXX at 195.112.214.99>;tag=as7b43dfbd'
-- SIP/callshop-3fcc is circuit-busy
== Everyone is busy/congested at this time
-- Got SIP response 481 "Call Leg Does Not Exist"
back from 213.61.187.150
My sip box is :195.112.214.99
The voip provider sip box:213.61.187.150
the configuration of my sip file was like this:
sip.conf
[callshop]
type=peer
host=213.61.187.150
username=XXXXX
secret=XXXXX
but when ive added these lines on my sip.conf file:
[callshop]
type=peer
host=213.61.187.150
username=XXXXX
secret=XXXXX
fromuser=XXXX
usereqphone=yes
canreinvite=no
nat=yes
insecure=invite
insecure=port
port=5060
disallow=all
allow=g729
it worked and the invitation problem was solved,but
when i have tried to send calls by iax to the same
voip provider the call failed like this:
dial 0017046872001 at calls
-- Executing Dial("OSS/dsp",
"IAX2/callshopcompany/0017046872001") in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format g729)
-- Format for call is g729
-- Hungup 'IAX2/callshopcompany/3'
== No one is available to answer at this time
Oct 16 20:35:53 WARNING[30499]: pbx.c:1949
ast_pbx_run: Timeout, but no rule 't' in context
'calls'
<< Hangup on console >>
My iax.conf file is like this:
[callshopcompany]
type=peer
host=sip.callshopcompany.com
username=875630553
secret=darWish472
the voip provider that iam dealing with accept both
sip and iax,but i realize in this case that something
must be added to the iax.conf file to solve the
invitation problem like i have done this in sip.conf
file.
Regards;
Jonny
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