On Fri, 3 Jun 2005, Rich Adamson wrote:
Thanks for the reply!
I just got back in town and am anxious to spend the entire weekend on
asterisk and will be trying this newly learned stuff out!
I will report back on how it goes!
Take care,
Steve
I'm going to try and ask this again and keep it short and as too the
point as I can while still providing enough info to be of use.
PLEASE advise if I am going about this wrong or asking too much.
I'm seriously doing my BEST to throughly read the docs and try a bunch of
things BEFORE coming here to ask and possibly annoy.
If is documentation that explains thsi process in terms that someone new
can grab onto quickly I'm missing it!
OK here goes again :-)
I have complied an asterisk system and got it going from scratch and all
works great except I cannot make an outbound sip-to-PSTN call and do not
fully understand how to configure it.
I've been folowing some examples and keep running into this stumbling
block:
As soon as I add (to sip.conf) this section:
[siprovider.com]
type=peer
host=sipprovider.com
fromuser=2135551212
secret=2135551212
authname=2135551212
fromdomain=siprovider.com
I no longer can recieve ANY inbound calls from the PSTN via my sip
provider.
I've tried many variations of attempting to get this section (I think it's
referred to a 'sip channel) into my sip.conf all which give the same
result.....
All inbound calls from PSTN TO this account FAIL.
I have tried with the dialplan in context [default]
with a test dialplan and with a 'blank' dial plan.
every way I try this, inbound calls via SIP and my SIP provider stop
reaching my asterisk box.
If I remove the above shown section leaving only the
register => 2135551212:[EMAIL PROTECTED]
all works great and calls come in from the PSTN to my asterisk box and
people can get around my menu just fine and dial internal SIP extension
numbers.
Let's see if I can at least help out with the understanding part (and
I'm doing this from memory, and not currently using any sip providers).
First, change your context name ([siprovider.com]) to something
different to avoid confusion as to that context name and the
host=sipprovider.com statement. Might try something like [sipprovider-com].
Ok can certainly do that to make it more clear.
The sipprovider.com _must_ resolve to a valid IP address using
DNS in your example above.
Certainly! and it does!
The register statement does nothing more then tell your provider
that your on line, and to use whatever is at the end of the
statement (/1234) as the extension number to execute in your dialplan
when sending you an inbound call. Since you have nothing at the end
of the register statement, your inbound calls must be processed via
the exten=>s approach (which you apparently are doing).
That is correct, and that part works VERY well and this is one thing where
I actually do understand what is going on :-)
I have also redirected it to other extensions and of course that works
well too.
At this point I'm going to cut short and go try some of the advice that
follows,
I will report back with most likely a very happy and working system that I
can make outbound calls on!
I GREATLY appreciate your time to reply and explain some of the concepts
that I was having difficulty grasping from the docs I have read so far.
Now to go give some of this a shot!
Take care!!!
Steve Gladden
--------------------
When dialing an outbound sip call (via your sip provider), the Dial()
statement can use the form:
exten => _1XXXXXXXXXX,1,Dial(SIP/myOutContext)
where the myOutContext would look something like:
[myOutContext]
type=peer ; for outbound calls
(other parameters as needed to authenticate an outbound call)
For inbound sip calls (via your sip provider), use a context something
like:
[myInContext]
type=user ; for inbound calls
context=InboundSip
(other parameters as needed to authenticate or qualify the inbound
call)
Look carefully at the list of valid parameters for type=user verses
type=peer in /usr/src/asterisk/configs/sip.conf.sample paying close
attention to what's listed in each colume. (Note: authname= is not
listed in the current sip.conf.sample file.)
With most itsp's, you should be able to process outbound calls by
simply using the exten => _1XXXXXXXXXX,1,Dial(SIP/myOutContext)
without the register. (This sort of varies by itsp though.) Try
it and debug that before going on to incoming calls.
Once outgoing calls are accepted, then add the [myInContext] section.
Look closely at the results from 'sip debug' when an inbound call
is placed. If there is a problem with context, authentication, etc,
the debug output will give you a pretty good clue what is not right.
Also, read the comments in the sample config file carefully. For
example: "For incoming calls only. Example: FWD (Free World Dialup)
We match on IP address of the proxy for incoming calls since we
can not match on username (caller id)"
You might find that your outbound calls are sent to one IP address
(host=) while inbound calls come from a different IP address. That's
not uncommon for any reasonable sized itsp. So, you may need multiple
contexts in sip.conf to handle that. In any case, 'sip debug' is
your friend.
Or, you can combine the above into a single type=friend context like:
[broadvoice] ; this is referenced for outgoing calls to Broadvoice.com
type=friend
username=3035551212 ; not needed as its in the Register statement
secret=x65xv1234z
host=sip.broadvoice.com
insecure=very
canreinvite=no
dtmfmode=inband
fromuser=3035551212
fromdomain=sip.broadvoice.com
context=from-broadvoice ; in extensions.conf
disallow=all
allow=ulaw
deny=0.0.0.0/0.0.0.0
permit=147.135.8.129/255.255.255.0
permit=147.135.0.129/255.255.255.0
permit=147.135.4.128/255.255.255.0
that includes elements from both type=user and type=friend, securing
(somewhat) you inbound calls by insisting they originate from selected
class-c networks owned by your itsp (permit=).
There really isn't any single way (or best way) to define the above
contexts as each itsp _may_ have different requirements. Keep in mind
that your itsp may not be using asterisk at all.
Rich
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