No subject
Thu Jul 12 09:23:04 CDT 2007
help me in another issue related also to registering
asterisk with another softswitch:
A) If nat=yes, then I have to set canreinvite=no to be
able to register, correct?
B) In case of using firefly softphone, how it possible
to set it to have nat=yes (at the firefly it self and
not at the sip user configuration section)? As most of
the sip endpoint give an option to set nat=yes and so
on, how it can be done with firefly softphone?
C) One time I succeed to register my asterisk on
another softswitch (sip registeration), but when I
routed calls via this IP Trunk, then the calls are not
deliver to the softswitch at all, and the error at
asterisk says that eveyone is bussy. I do not know
why? Registeration succeed but calls are not appear at
all on the softswitch screen. By the way: my Asterisk
still does not support G711 while the softswitch that
I am attempting to register with it support only G729
and G723, is that the reason that the call does not
appear on the softswitch (after registeration
complete)? Normally on that softswitch, when endpoints
are registering and they dot match the codec, then
calls are delviered and it appears on the softswitch
but it gives a message that codec miss match, but in
my case it does not even display and kind of receiving
the call from asterisk with fail or without, any
advise? Is it because my softswitch does not support
G711? I beleive it should process the call with fail
(codec miss match), but I do not see the call.
Looking to hear from you.
Regards
Bilal
--------------------------------------------
Bilal,
On Tue, 23 Oct 2007, bilal ghayyad wrote:
> This is if I need to let Asterisk register with
another softswitch
(so I
> used register =>), what if I need asterisk to send
call for the
> softswitch without register to it (directly)? If I
removed the
register
> => then how it will distiguish the IP address in the
"host" at
the
> [sip_trunk] is the IP address of the softswitch that
need to
register
> with it and not the IP address of the original
caller sip
endpoint?
Unless I am missing something here, I suppose the
answer is that
Asterisk can distinguish the IP endpoints because they
are ...
distinct.
Here is the essence of the situation:
In Asterisk it is possible to peer with an endpoint
with and without
registrations. Registrations are mostly intended for
dynamic endpoints
whose IP address can potentially change, such as
end-user phones off of
broadband connections, or other clients whose IP
address is not
desirable
to track or cannot be trusted.
The other type of peer is a 'trusted' trunk tied to
a particular IP
endpoint on the far end. The trust can be done only
by IP address,
or by IP address + SIP UDP port. This type of peer is
typically used
when doing SIP handoff from origination and
termination carriers on any
kind of large-scale, or in other intra-industrial
and/or internal
and/or
intra-platform SIP connections where it is not
desirable to position
one
endpoint of the SIP trunk as a UAC (client)
registering against a UAS
(server) per se, as such, in the respect that one
challenges the other
for authentication credentials.
So, what I would do is build a trusted trunk
(type=peer,
insecure=very)
to the softswitch that has a static IP (host=)
endpoint defined. Then,
Asterisk can accept registrations from your users.
Where to route the
call is determined entirely in the dial plan
(extensions.conf), where
you can send calls to particular SIP peers. So, for
example, here is a
regular user defined in sip.conf:
[Alex_Evariste_2]
type=friend
host=dynamic
canreinvite=no
username=Alex_Evariste_2
secret=xxxxxx
nat=yes
allow=ulaw
qualify=yes
mailbox=1000 at evariste
context=default-user-dial
And here is a dedicated trunk to a provider:
[my_sip_provider]
host=xxx.yyy.zzz.www
insecure=very
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833
Then, your dial plan for a user can be set up like
this, for
example,
in extensions.conf:
[default-user-dial]
; Any North American ten-digit number.
exten =>
_NXXXXXXXXX,1,Dial(SIP/${EXTEN}@my_sip_provider)
In our case, we actually register with our SIP
origination provider,
so
we have this IP trunk:
[junction_networks]
fromdomain=jnctn.net
host=sip.jnctn.net
port=5060
insecure=very
username=this_user
secret=this_password
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833
But in addition, in the [general] context at the
top of sip.conf, we
have:
register => our_user:our_password at sip.jnctn.net
As you can see, one type of registration
requirement does not
interfere
with another.
Hope this helps. If it doesn't, please let me know
if I
misunderstood
something.
Cheers,
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671
P.S.
On Tue, 23 Oct 2007, Alex Balashov wrote:
> [junction_networks]
>
> fromdomain=jnctn.net
> host=sip.jnctn.net
> port=5060
> insecure=very
> username=this_user <--
> secret=this_password <--
> type=peer
> qualify=no
> canreinvite=no
> dtmfmode=rfc2833
The 'username' and 'secret' there are actually not
required unless
we
were to challenge Junction in other direction, which
would be
impossible
with a trunk defined as 'insecure=very' anyway. But
since we receive
no
calls from them, it is completely unnecessary in every
respect. Not
sure
why we have it.
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671
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