asterisk users - Jun 2005 - CLUELESS NEWBIE needs help making an outbound sip call to PSTN

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 => 2135551212:2135551212@sipproviderexample.com

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.

This of course leaves me with no SIP [brackted] section of which to use 
for outbound calls of which I'd love to eventually get working.

Am I doing this right at all???? or am I headed completely in the wrong 
direction here?

I also tried this with a FREE Stanaphone account and get a very similar 
but strange result....

IN both cases adding this section to sip.conf result in my calls 
terminating at the SIP provider voicemail system instead of coming into my 
asterisk box here.



A side note:
Not that it really matters but here's what I get from my provider if I try 
to dial into my PSTN number from the PSTN:

Standard unavailable voicemail message as if not registered in.

-and-
>From Stanaphone:
a 'strange' voice message that gives you the option to either #1 change
my
outgoing unavailable message or #2 press ANY key besides #1 to hangup.

Any help or pointers would be GREATLY appreciated!!!!!


I compiled and Installed Asterisk about 10 days or so ago and am running
version:
CVS-Nv1-0-7-05/19/05-11:22:20 built by root@Vontage on a i686 
running Linux

Thanks!!!

Having a BLAST!


Steve Gladden

> 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.
>
> Steve Gladden
To get an example setup for your provider, try searching google:
<provider> site:lists.digium.com

If you go onto your provider's website, they usually have an example
configuration for you.

If you can't find an example that works, post the sip.conf entry, the
appropriate section of extensions.conf, and the console output.

Kevin

> 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 => 2135551212:2135551212@sipproviderexample.com
> 
> 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].
The sipprovider.com _must_ resolve to a valid IP address using
DNS in your example above.

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).

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

Finally got it to work with my provider,
The task was seemingly a simple one which was to (simply) be able to make
an outbound call to the PSTN via our sip provider
Unfortunately none of the information that I was given seemed to have any 
effect on the problems I was having...
However it was helpful and provided needed insight along with the 
opportunity to get into CVS HEad a bit and so some compiling.

I VERY much appreciate the help but I still find that I did not learn as 
much as would have like have but will 'keep trying'!!!

Just ended up through very much tral and error pasting other people's 
configurations in until finally one worked.


The one I got to work with Stanaphone did not work with my provider,

I found one of (many) that someone had listed for broadvoice that worked
with my provider (who is *NOT* broadvoice) :-)


Unfortunately I am not knowledgable on what is exactly at my sip 
provider's end and they are not willing to disclose with me or this list
what that hardware / software combo on their end (the provider) is.

Here is my sip.conf that FINALLY worked after about 3 weeks of trying 
things.



Steve


;-------------Testing------------------


[general]
  port = 5060
   bindaddr = 0.0.0.0
    allow=ulaw
  ;  dtmfmode=info
  ;  nat=yes



     ; This section is because i'm behind nat
      externip = x.x.x.x ;Outside address
       localnet = 10.73.73.133 ;Inside address
        localmask = 255.255.255.0 ;Inside subnet

         context = sip ; Default context for incoming calls
          register => ##########:secret@sip.stanaphone.com/1000
          register => ##########:secret@sip.provider.net/4078
          register => ##########:secret@sip.provider.net/4077


[stanaphone-out]

;works!!!
host=sip.stanaphone.com
context=sip
type=friend
dtmfmode=rfc2833
canredirect=no
disallow=all
allow=ulaw
insecure=very
username=secret
fromuser=secret
secret=secret


;more testing broadvoice examples
;THIS ONE WORKS!!!

[our-sip-provider-out]
type = peer
host = sip.provider.net
secret = secret
user=phone ; I needed this to make it work (what tha ????)
fromuser = secret
username= secret
authname= secret
fromdomain = sip.provider.net
context = sip
insecure=very ; To allow registered hosts to call without 
re-authenticating
canreinvite = no
; BV claims they support rfc2833, but for some reason passing digits
; after a connected call only works with inband
dtmfmode = rfc2833
;dtmf=inband

CVS-HEAD
Running Version:
Asterisk CVS-HEAD built by root@Vontage on a i686 running 
Linux on 2005-06-06 22:32:05


*CLI> show version files
File                      Revision
----                      --------
cdr_custom.c              Revision: 1.11
cdr_manager.c             Revision: 1.6
cdr_csv.c                 Revision: 1.16
pbx_functions.c           Revision: 1.3
chan_zap.c                Revision: 1.458
chan_phone.c              Revision: 1.52
chan_modem_i4l.c          Revision: 1.27
chan_oss.c                Revision: 1.49
chan_features.c           Revision: 1.12
chan_skinny.c             Revision: 1.78
chan_local.c              Revision: 1.47
chan_iax2.c               Revision: 1.303
iax2-parser.c             Revision: 1.45
iax2-provision.c          Revision: 1.12
chan_mgcp.c               Revision: 1.123
chan_agent.c              Revision: 1.136
chan_modem_bestdata.c     Revision: 1.16
chan_sip.c                Revision: 1.754
chan_modem_aopen.c        Revision: 1.15
chan_modem.c              Revision: 1.40
io.c                      Revision: 1.10
sched.c                   Revision: 1.19
logger.c                  Revision: 1.74
frame.c                   Revision: 1.57
loader.c                  Revision: 1.45
config.c                  Revision: 1.66
channel.c                 Revision: 1.202
translate.c               Revision: 1.37
file.c                    Revision: 1.68
say.c                     Revision: 1.60
pbx.c                     Revision: 1.254
cli.c                     Revision: 1.86
md5.c                     Revision: 1.14
term.c                    Revision: 1.10
ulaw.c                    Revision: 1.4
alaw.c                    Revision: 1.3
callerid.c                Revision: 1.32
fskmodem.c                Revision: 1.7
image.c                   Revision: 1.15
app.c                     Revision: 1.66
cdr.c                     Revision: 1.40
tdd.c                     Revision: 1.6
acl.c                     Revision: 1.45
rtp.c                     Revision: 1.133
manager.c                 Revision: 1.99
asterisk.c                Revision: 1.162
dsp.c                     Revision: 1.43
chanvars.c                Revision: 1.8
indications.c             Revision: 1.25
autoservice.c             Revision: 1.12
db.c                      Revision: 1.18
privacy.c                 Revision: 1.5
enum.c                    Revision: 1.26
srv.c                     Revision: 1.13
dns.c                     Revision: 1.14
utils.c                   Revision: 1.47
config_old.c              Revision: 1.4
plc.c                     Revision: 1.5
jitterbuf.c               Revision: 1.15
dnsmgr.c                  Revision: 1.5


Sorry for the LONG delay on this wrap up.


Take care!

Steve