Ok I'm still playing and the way it's supposed to work is making much more
sense now.
However it is still 'not working' as soon as I add this:
[sipproviderexample.com]
type=peer
host=sipprovider.com
fromuser=2135551212
secret=2135551212
fromdomain=sipproviderexample.com
to sip.conf
I also learned that I needed to replace the internal IP host= with
sipproviderexample.com
for whatever reason the example I had been working off of was showing
an internal address for that
What is breaking is that the asterisk box stops accepting ANY inbound
calls via sipprovider.com as soon as I add the extension noted above.
I have also tried many variations of that...
and using another context besides default.
No change.
It flat out 'stops working' no inbound calls (from PTSN via sip provider)
as soon as I add that to sip.conf and associate it with any ANY context.
even associating it with an empty context (no dialplan lines) in
extensions.conf it behaves the same way.
If I comment it out inbound calls work perfectly well all day long
I'm baffled and it's very frusterating!
if I attempt an inbound call in this non-working state from PSTN this is
what I see with sip debug:
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKdf72.63b2d483.1
From: "unknown"
<sip:[EMAIL PROTECTED]:5060;user=phone>;tag=e2d90014-1cffac42-8cd690d8
Call-ID: [EMAIL PROTECTED]
To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as51e3a2ab
CSeq: 1131072 ACK
User-Agent: SIPProxy
Content-Length: 0
---------> About 5 seconds later this comes back
8 headers, 0 lines
Destroying call '[EMAIL PROTECTED]'
remove those few lines, reload and incoming works just fine again.
I also tried this with a different provider (stanaphone) and that behaves
the same way.
Anything else to try?
Thanks for all the help and this is EXTREMELY cool! I am playing and
having a blast wit hit....
Sure would like to be able to make some outbound calls! :-)
Take care!
Steve
On Thu, 2 Jun 2005, Ronald Wiplinger wrote:
Steve wrote:
I have read LOTS of docs and played quite a bit to get this far....
Good, keep playing!!!
(a lot of your typing time deleted)
----------------------------
OK here's what messes it all up (and I admit I'm clueless here)
register => 2135551212:[EMAIL PROTECTED]
[sipproviderexample.com]
type=peer
host=10.77.77.133
fromuser=2135551212
secret=2135551212
fromdomain=sipproviderexample.com
adding this secttion breaks it and I really do not understand what it's
even
for...
What does it mean for you, that this "breaks" it. Did it work before?
What is your "jump in point" to the dial plan? You do not have a context=
line. So you may jump into the context=default as usually mentioned in
sip.conf
[general]
context=default ; Default context for incoming calls
Do you have something in the dialplan like:
[from-sipproviderexample.com] ; if you would use this as your
context= in sip.conf, otherwise the next lines in the default context of your
extensions.conf
exten => _X.,1,Dial(SIP/601,60,tr) ; if any number calling from this
provider should ring extension 601
or
exten => 2135551212 ,3,Dial(SIP/601,60,tr) ; if only a dialed in number
2135551212 of your provider should dial extension 601
Keep playing with:
context=xxxxx and [xxxxx]
exten => something with and without starting _
Hope it helps, ...
bye
Ronald
does it work with the register line somehow? or is it totally seperate?
what is it for?
All the docs I have looked at seem to suggest adding this extra section but
do
not really seem to explain it or what exactly it does.
I'm not sure what it's for or if it has anything to do with making outbound
sip
calls from the internal extensions.
when I add it my sip provider account stops working and I get registration
retries and timeouts without any successful registrations after that.
I'm just looking for a good pointer in where to go for an example of how to
use
my provider account for outbound connections...
I understand the dialplans themselves but do not know how to associate
them
with the actualy sip provider account for an outbound call.
Thanks....
I'll keep reading until I figure these things out but any pointers to
specific documentation that answers any of these questions would be very
much
appreciated....
I *think* I am familiar with just about all of the standard asterisk
documentation I have been able to find...
More than likely I am just missing some key points that I have read but
have
misinterpreted!
Take care!
Steve
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