[Serusers] Call Transfers in SER + Asterisk architecture

[Walter Rodrigues Filho]

Ricardo -
 
this is an Asterisk issue.
 
Configure your SIP.conf in * creating a sip peer with insecure=very fotr the
SER peer, like this:
 
[ser-peer]
type = friend                  ; should you want to receive and make calls
to SER
context=from-ser            ; the context in dialplan with extensions
allowed to be accessed by SER. Here you must have PSTN extensions
capability.
host=200.XXX.XXX.XXX    ; the IP address for SER
fromdomain=domain.com ;the Domain part of uri to be verified by asterisk on
the INVITE received by SER. 
qualify= yes                        ; just to check the latency between SER
and Asterisk (like this, if over 2000ms Ast will report as unavailable
peer).
disallow=all
allow=alaw
allow=g729
insecure= very                ; this line garantees that any username part
of Request URI sent by SER in INVITE to Asterisk will be accepted by Ast and
routed to the dialplan.
 
So, if SER send an INVITE to HYPERLINK
"mailto:5531332818847 at domain.com.br"5531332818847 at domain.com.br , Asterisk
will look for a section of type =user in SIP conf to match the user part
first, it won't find, then it will look for a type=peer. It will find and
try to match the IP address as in host= ...line and will accept any username
part as per insecure=very line. Then, if context=from-ser in the Ast
dialplan allows this dialing string (553132818847), it will proceed from
there.
 
Hope it helps.
 
At.
Walter

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