Hey Mike,
Thanks for the information. Fortunately, I have that book from
www.voip-info.org/digium form and that is the source that made me able to call
from my sip phone to my asterisk server but not able to go beyond then
that and
for that you helped me here. I found that book like to the point and just
perfect i must say.
Thaks a lot for your fruitful information. Now, I am dealing with another
problem here. There is a cisco call manager currently installed in my
lab setup
by my network administrator and now i will ahve to connect my pbx with the
cisco call manager and just talked to my administrator and he explained
something about that which might hinder my progress. So, let's see if he can
help me with this otherwise i will tell you about this but before that i will
do some google. :) :) :) :)
My current set up is like this:
sip/Iax soft and hard
phones(sourcess)----AsteriskPBX----Ciscoswitch----CiscocallManager---Service
Server----BELL/ROGERS/Telus phone service company---Sip/IAX soft and hard
phones(destinationss)
Question: Can cisco call manager transer multiple calls generated by one
AsterixPBX to the destinations in my above setup?
I know now that the direct connection of my AsteriskPBX to the Service server
without cisco call manager will work excellent but there is Another
Ciscocallmanager which I am sure will distrub calls.
The main thing no body will allow me to get the access of cisco call
manager as
if i mess that up then our department set up might ruined so i doing research
before going for anything hazardous situation.
Cheers,
Samir.
Quoting "Mike C. Fletcher" <[EMAIL PROTECTED]>:
Samirkumar Patel wrote:
Currently, I am using VoIP service from rogers here in School lab.
I can make calls from my sip and iax soft phone to asterisk server by some
preconfigured numbers ( like 100,1,2000, etc. ) but I have followign
concerns.
How can I connect my asterisk server to my VoIP service provider
network(VoIP
gateway) to reach to the outside world ?
Do I need to ask to my VoIP service provider for that ? for username and
password because I am not administrator here and being a student I
am not aware
about that. :) I will have to ask to my network administrator. :)
Asterisk can use a SIP account as an incoming and outgoing channel.
I've only personally set it up for incoming, but I would guess it's a
very similar setup for outgoing (at least, that's what the comments
in the file seem to say). The same parameters you are using to
connect your soft-phone to your service provider will allow you to
have your asterisk server connect as a SIP phone to the service. (If
you mean you don't have a SIP username and password, well, you'll
have to get them). See the sip.conf file in the /etc/asterisk
directory for information on how to do it.
Now, encoding a password into the config file is sub-optimal, of
course, but it does work (I believe there may be a way to define the
password elsewhere, but I haven't looked into it). For an academic
demo this is probably sufficient, just be sure to protect the file
from prying eyes. My impression is that using the SIP channel instead
of an IAX one is largely deprecated as a "real world" connection
mechanism (IAX being far preferable), but again, for an academic
demo, or in special circumstances, it does seem to work.
Again, lots of instructions in the sip.conf file. You may also find
that reading "Asterisk the Future of Telephony" (available free,
online)[1] helps you with lots of these "where do I put my finger"
questions. It takes about 6 hours to read cover-to-cover, and gives
a good general background for understanding how asterisk works, what
you can do with it, and where to look to figure out how to do those
things. Quite a reasonable investment in time.
HTH,
Mike
[1] http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
--
________________________________________________
Mike C. Fletcher
Designer, VR Plumber, Coder
http://www.vrplumber.com
http://blog.vrplumber.com
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