28 Apr
2005
28 Apr
'05
2:42 p.m.
Hi
I originally had a setup like this, but to me it just seemed way to
expensive to be doing this, since you will need to add asterisk boxes,
just to keep doing these "u-turns", and because you want the billing
info. Also, I started to have problem when I hit the external gateway,
because originally again i did this via asterisk, which then passing
data back to ser gave headaches. But if anyone gets this combo up and
running weel would be interested to know complete setup.
Iqbal
Kamran Ahmad wrote:
hello Carles Wang
This is the way to to successfully call through
asterisk.
SER is Main Registrar. Asterisk is B2BUA.
SER will route its call to asterisk.
now control is at asterisk side he can do any thing to
that call (forword, voicemail, etc....). but actually
asterisk dont know who is the user he ll route this
call back to SER who is aware of the user.
it is like U-Turn through Asterisk. try this and tell
me
ser.cfg
------------------------
rewritehostport("asterisk:port");
force_rport();
fix_nated_contact();
force_rtp_proxy();
t_relay();
sip.conf
------------------------
[general]
context=default
port=5070
bindaddr=0.0.0.0
srvlookup=no
[2000]
callerid=2000
type=friend
host=dynamic
canreinvite=no
reinvite=no
qualify=yes
auth=plaintext
nat=yes
[3000]
type=friend
host=dynamic
callerid=3000
qualify=yes
canreinvite=no
reinvite=no
auth=plaintext
nat=yes
--- Charles Wang <lazy.charles(a)gmail.com> wrote:
I also have the same problem. I don't know how
to
fix it. Do you get
any solution or sample config about it. I think it
is a problem of
asterisk.
On 4/28/05, Kamran Ahmad <p_kami(a)yahoo.com> wrote:
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hi list
any person using ser and Asterisk as B2bUA.
what is the best way of using asterisk as B2BUA.
i can route calls to through asterisk its working
but
some time one UA tries to hangup the call and the
other side is not able to receive Bye
what could be the best way of using asterisk for
monitering voice
Thanks
Kamran
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--
Best Regards
Charles