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Okay, since VoIPDiscount have started charging, I have had another play

Success!!

My configurations is as follows...

Asterisk@home 1.5 running on a local IP behind a router.

The following ports are forwarded to the asterisk box
5060 (UDP)
8000->8010 (UDP)
4560->4570 (UDP)
4560->4570 (TCP)

------------

In my sip.conf I have the following extra lines (Maintanace, Config Edit,
sip.conf)

externip=MY ROUTERS EXTERNAL IP ADDRESS or host name if you have one
localnet=192.168.1.0/255.255.255.0 (change this to suit your network)
canreinvite=no

---------

If something is not mentioned, it is either blank, or at it's default
setting
If something is in (round brackets), it is for information and should be
replaced with your details, or omitted.

Trunks -> sipdiscount

Maximum Channels - 1

Dial Rules - 0044+XXXXXXXXXX
004420+XXXXXX (Replace 20 with your area code, minus the
leading 0)
0044+800.
0044+808.

Trunk Name - sipdiscount

PEER Details -
allow=ulaw&alaw
authuser=(your username)
disallow=all
fromdomain=sipdiscount.com
fromuser=(your username)
host=sip.sipdiscount.com
insecure=very
nat=yes
qualify=yes
secret=(your password)
type=peer
username=(your username)


Everything else blank

--------------------
Outbound Routing -> sipdiscount

Dial Patterns
00441.
00442.
00443.
0|800.
0|808.
0|1.
0|2.
0|3.
[2-8].


Trunk Sequence
SIP/sipdiscount

-----------------

With this setup, you can just dial UK land lines & freephone numbers via
sipdiscount like a normal phone (no prefixing required!)


Hope this helps!
Sparks....

The subject should, of course, read..
SIP Discount with Asterisk - Success!

Doh!

Sparks wrote:
|| Okay, since VoIPBuster have started charging, I have had another
|| play
||
|| Success!!
||

I just get "All Circuits Are Busy Now"

>
> Sparks wrote:
> || Okay, since VoIPBuster have started charging, I have had another
> || play
> ||
> || Success!!
> ||
>
> I just get "All Circuits Are Busy Now"

Still working OK for me now!

One ammendment, in the Trunks -> sipdiscount

The line "004420+XXXXXX"
should be "004420+XXXXXXXX"

I have also found, if you set the "Maximum Channels" to more than 1, you can
make multtiple calls at the same time

Sparks...

Sparks wrote:
||| Sparks wrote:
||||| Okay, since VoIPBuster have started charging, I have had another
||||| play
|||||
||||| Success!!
|||||
|||
||| I just get "All Circuits Are Busy Now"
||
|| Still working OK for me now!
||
|| One ammendment, in the Trunks -> sipdiscount
||
|| The line "004420+XXXXXX"
|| should be "004420+XXXXXXXX"
||
|| I have also found, if you set the "Maximum Channels" to more than 1,
|| you can make multtiple calls at the same time
||
|| Sparks...

Encouraging.............

I picked up on the missing X, though not living in the London area, the
number of Xs were right for me!

Anyway, where exactly did you put the three extra lines in your .conf file?

Could you post your entire sip.conf file? (personal stuff deleted, of
course)

when you say ammend 192.168.1.0 to suit, what did you mean? Is this your
gateway address?

I can't even get the Test login to work!

> Encouraging.............
>
> I picked up on the missing X, though not living in the London area, the
> number of Xs were right for me!
>
> Anyway, where exactly did you put the three extra lines in your .conf
> file?
>
> Could you post your entire sip.conf file? (personal stuff deleted, of
> course)
>
> when you say ammend 192.168.1.0 to suit, what did you mean? Is this your
> gateway address?

Nope, this is your local network address.

So for example, if all your PC's on your network were in the 10.12.25.X
range, you would use localnet=10.12.25.0/255.255.255.0

If you PC's were in the 10.12.X.X range you would use
localnet=10.12.0.0/255.255.0.0

If you are on and ADSL connection with a dynamic address, it is best to
register with a service like www.no-ip.org
they will (for free) provide you with a DNS name (like whatever.no-ip.org)
then you use this in your externip= line.
You then need to install a cluent that will update this DNS address if your
IP address changes (otherwise you will need to change the externip= line
every time your IP address changes) (See www.no-ip.org for sutable clients)

> I can't even get the Test login to work!

Okay, here is the entire sip.conf...

-----------------
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
externip=XXX.XXX.XXX.XXX ; Change this to your router's IP address, or your
DNS name
localnet=192.168.1.0/255.255.255.0
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
canreinvite=no

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
----------------

"Sparks" <postmaster@127.0.0.1> wrote in message
news:435f4315$0$38038$.. .
>
>> Encouraging.............
>>
>> I picked up on the missing X, though not living in the London area, the
>> number of Xs were right for me!
>>
>> Anyway, where exactly did you put the three extra lines in your .conf
>> file?
>>
>> Could you post your entire sip.conf file? (personal stuff deleted, of
>> course)
>>
>> when you say ammend 192.168.1.0 to suit, what did you mean? Is this your
>> gateway address?
>
> Nope, this is your local network address.
>
> So for example, if all your PC's on your network were in the 10.12.25.X
> range, you would use localnet=10.12.25.0/255.255.255.0
>
> If you PC's were in the 10.12.X.X range you would use
> localnet=10.12.0.0/255.255.0.0
>
> If you are on and ADSL connection with a dynamic address, it is best to
> register with a service like www.no-ip.org
> they will (for free) provide you with a DNS name (like whatever.no-ip.org)
> then you use this in your externip= line.
> You then need to install a cluent that will update this DNS address if
> your IP address changes (otherwise you will need to change the externip=
> line every time your IP address changes) (See www.no-ip.org for sutable
> clients)
>
>> I can't even get the Test login to work!
>
> Okay, here is the entire sip.conf...
>
> -----------------
> ; Note: If your SIP devices are behind a NAT and your Asterisk
> ; server isn't, try adding "nat=1" to each peer definition to
> ; solve translation problems.
>
> [general]
>
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> disallow=all
> externip=XXX.XXX.XXX.XXX ; Change this to your router's IP address, or
> your DNS name
> localnet=192.168.1.0/255.255.255.0
> allow=ulaw
> allow=alaw
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
> canreinvite=no
>
> #include sip_nat.conf
> #include sip_custom.conf
> #include sip_additional.conf
> ----------------
>
>
>

Hi Sparks,

Thanks for this, I'll check out your suggestions.

I already have a dyndns address.

"Sparks" <postmaster@127.0.0.1> wrote in message news:435f4315$0
>
> ; Note: If your SIP devices are behind a NAT and your Asterisk
> ; server isn't, try adding "nat=1" to each peer definition to
> ; solve translation problems.
>
> [general]
>
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> disallow=all
> externip=XXX.XXX.XXX.XXX ; Change this to your dyndns name
> localnet=192.168.1.0/255.255.255.0
> allow=ulaw
> allow=alaw
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
> canreinvite=no
>
> #include sip_nat.conf
> #include sip_custom.conf
> #include sip_additional.conf
> ----------------

Hi Sparks,

Here's my sip.conf - I can't see anything different (can you?)

I've got all the same ports as you forwarded in the router.............but
it still doesn't work

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
externip=my.dyndns.address
localnet=192.168.1.0/255.255.255.0
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
canreinvite=no

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

> Hi Sparks,
>
> Here's my sip.conf - I can't see anything different (can you?)
>
> I've got all the same ports as you forwarded in the router.............but
> it still doesn't work
>
> [general]
>
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> disallow=all
> externip=my.dyndns.address
> localnet=192.168.1.0/255.255.255.0
> allow=ulaw
> allow=alaw
> context = from-sip-external ; Send unknown SIP callers to this context
> callerid = Unknown
> canreinvite=no
>
> #include sip_nat.conf
> #include sip_custom.conf
> #include sip_additional.conf

Looks fine to me!

Have you reloaded the config files in asterisk since you made changed?
(Easiest thing to do is go to a trunk and press save, then press the red
"You have made changes - when finished, click here to APPLY them" at the top

Sparks...

"Sparks" <postmaster@127.0.0.1> wrote in message
news:435f6597$0$38039$.. .
>> Hi Sparks,
>>
>> Here's my sip.conf - I can't see anything different (can you?)
>>
>> I've got all the same ports as you forwarded in the
>> router.............but it still doesn't work
>>
>> [general]
>>
>> port = 5060 ; Port to bind to (SIP is 5060)
>> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
>> disallow=all
>> externip=my.dyndns.address
>> localnet=192.168.1.0/255.255.255.0
>> allow=ulaw
>> allow=alaw
>> context = from-sip-external ; Send unknown SIP callers to this context
>> callerid = Unknown
>> canreinvite=no
>>
>> #include sip_nat.conf
>> #include sip_custom.conf
>> #include sip_additional.conf
>
> Looks fine to me!
>
> Have you reloaded the config files in asterisk since you made changed?
> (Easiest thing to do is go to a trunk and press save, then press the red
> "You have made changes - when finished, click here to APPLY them" at the
> top
>
> Sparks...
>

Hmm, I have, yes.

Perhaps you would be kind enough to post the relevent section of your
sip_additional.conf file?

Mine's:

[locsipdisc]
username=MyUsername
type=peer
secret=MyPassword
qualify=yes
nat=yes
insecure=very
host=sip.sipdiscount.com
fromuser=MyUsername
fromdomain=sipdiscount.com
disallow=all
authuser=MyUsername

Cheers.