[Asterisk-Users] iax invtation problem

[Juan Janczuk]

-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]En nombre de jonny
hashem
Enviado el: Domingo, 16 de Octubre de 2005 02:44 p.m.
Para: asterisk-users at lists.digium.com
Asunto: [Asterisk-Users] iax invtation problem


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

John:
Are you SURE that your provider is accepting G729?
I've seen that error (== No one is available to answer at this time), when
the * box had no codec to use.
As far I can see, you only have G729 as allowed codec.....

Hope this helps.
Juan.





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