[kash]

i've been messing around with trying to get my pstn into my asterisk box but i cant seem to figure it out.. i tried to use the config wizard which is great .. but i'm using asterisk at home so i have to go though amp to edit the conf's but thats where i'm getting confused does anyone have a how to get teh sipura working with asterisk and amp please

i did get my line1 phone to work as an extension with asterisk but i just cant get the pstn to work as a trunk

tia

[jmattwood]

This is what's working for me - in Amp, create a new SIP trunk & call it PSTN-SPA3K

Under Outgoing settings:

auth=md5
context=from-internal
dtmfmode=rfc2833
fromuser=asterisk ;this user is created with the SPA3000 wizard
host=192.168.1.7 ;this is the IP address of your SPA3000
insecure=very
port=5061 ;this is the PORT your Sipura PSTN tab is set to
secret=xxxx ;password entered in SPA Wiz
type=peer
username=asterisk

Under Incoming settings:

User Context - from-internal

User Details:

allow=ulaw
context=from-internal
disallow=all
dtmfmode=rfc2833
host=dynamic
insecure=very
secret=xxxx ;password created with SPA Wizard
type=friend

When I was first running the wizard, I was never sure which extension number to use, nor which password - all my extensions had the same password, which made setting up easier.

[RMHunter]

I'm having a problem sort of like this described between the asterisk and SPA3000..

I used the wizard to config the SPA3000... great tool

so here is the result:

inbound PSTN calls go to the proper * extension....
the line1 acts as a normal extension within *
(I can dial a sip phone on asterisk from the 'line1' handset and vice versa)
and I can dial out the PSTN from the 'line1' handset (using the # prefix)

what I can't do is dial out the PSTN from another sip phone connected to asterisk...

any ideas?

[jmattwood]

Have you created the PSTN-SPA3K trunk?

Have you pointed any outbound routing to the trunk?

[craig]

There are 3 different configs that are generated by the wizard for your sip.conf: 1) a Line1 entry, 2) a PSTN Line entry, and 3) a VoIP to PSTN entry. The first two are SIP clients and the 3rd is an outbound trunk, all on asterisk. For the functionality you've described, you are making a call via asterisk to the the SPA3000 trunk, which verifies the caller (asterisk) as configured in 3) and allows a bridge to your PSTN port (FXO).

Remember that the 3000 logically has a number of interfaces, namely FXS, FXO, Line 1 (VoIP 1) and PSTN Line (VoIP 2). Each one is indepenent and needs to be linked to the others, depending on your configuration needs.

From your comments, 1) is definatley working, since you can call between SIP phones and Line 1 (you will see that the Line 1 is "registered"). You should make sure that 2) is also registered, although this is for incoming PSTN calling and not the other way around. The key to your issue is to verify that the VoIP to PSTN Gateway functionality is configured as per the wizard. This is the required config to allow the bridge to occur in the SPA3000 and dial out.

Finally, your dial asterisk dial plan needs to be set up to dial out for a specific dialing pattern via the PSTN Line. This is the outboung routing to the PSTN-SPA3K trunk mentioned by jmattwood. This is held in extentions.conf. An example of stanard US 7 digit dialing from the wizard is:

exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _XXXXXXX,2,Congestion

The entry "pstn-spa3k" is 3) above in sip.conf and is the "trunk" mentioned earlier.

I would suggest look at the asterisk console output to get an idea of what's going on (start asterisk with 'asterisk -vvvc'). It will show the PSTN Line being registered and also show the call state of the SIP to PSTN Line call as it takes place. It should give you a lot of pointers.

Good luck!

[RMHunter]

So, then if I understand correctly...

Beyond the entries in sip.conf...

[350]
type=friend
host=dynamic
context=home
secret=somepassword
mailbox=350
dtmfmode=rfc2833
disallow=all
allow=ulaw

[351]
; If you're using Asterisk, this goes into the Incoming settings
; For your Trunk
type=friend
host=dynamic
; If using Asterisk@home, change the below line to context=from-internal
context=home
secret=somepassword
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

[pstn-spa3k]
; If you're using Asterisk, this section goes into the Outgoing Settings
; for your trunk.
type=peer
auth=md5
host=212.51.214.130
port=5061
secret= somepassword
username=asterisk
fromuser=asterisk
dtmfmode=rfc2833
; If using Asterisk@home, change the below line to context=from-internal
context=home
insecure=very

And the entries in extensions.conf....
[home]

exten => 350,1,Ringing
exten => 350,2,Dial(SIP/350,20,T)
exten => 350,3,Voicemail(u350)
exten => 350,4,Hangup

exten => 911,1,Dial(SIP/911@pstn-spa3k,60,)
exten => 911,2,Congestion

exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _XXXXXXX,2,Congestion

exten => _1800XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1800XXXXXXX,2,Congestion

exten => _1888XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1888XXXXXXX,2,Congestion

exten => _1877XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1877XXXXXXX,2,Congestion

exten => _1866XXXXXXX,1,Dial(SIP/${EXTEN}@pstn-spa3k,60,)
exten => _1866XXXXXXX,2,Congestion

I also need to register the adapter in the [general] section of sip.conf???

something like:

register => 351@pstn-spa3k:somepassword.



Thanks for the help.

[craig]

[350] and [351] are the extentions for your Line 1 and PSTN Line, I would assume. The imporant one for our discussion is the [pstn-spa3k] entry for the outgoing PSTN calls. The config looks fine (to my rookie eyes) and I assume that 212.51.214.130 is the IP address of the spa3k.

I don't have any registration statement in my sip.conf for this link, since each time I dial out it pulls the values from the [pstn-spa3k] section. Plus, the SPA itself isn't expecting to maintain a registration. It waits for an incoming VoIP call and then checks against the VoIP to PSTN Gateway authentication fields.

Is it still giving you problems? What does the asterisk console say? The asterisk console should show you a failed connection attempt if it's misconfigured. The next area to look at is the SPA config for the gateway function.

[RMHunter]

Yeah, still giving me problems...

If I attempt a call from either a SIP phone connected to asterisk OR from the telephone connected to line1, I get the same result..

example..

I'll dial 18005551212

*Consol says:

(when using a sip hardphone)
Executing Dial("SIP/300-36b6", "SIP/18005551212@pstn-spa3k|60|") in new stack
Called 18005551212@pstn-spa3k

(when using telephone connected to line1)
Executing Dial("SIP/350-2193", "SIP/18005551212@pstn-spa3k|60|") in new stack
Called 18005551212@pstn-spa3k


The result over the line is silence for both cases.

If I dial from the phone connected to line1 with a # pre-appended, (avoiding asterisk) the call goes through.

the 3k's software version is 2.0.11 and hardware version 2.0.1

[craig]

I just tried dialing out myself from a SIP softphone and get the same first line as you, but it then continues. Here is my output:

Executing Dial("SIP/craig-b8d1", "SIP/01234567890@pstn-spa3k|60|") in new stack
-- Called 01234567890@pstn-spa3k
-- SIP/pstn-spa3k-5d4b is ringing
-- SIP/pstn-spa3k-5d4b answered SIP/craig-b8d1
-- Attempting native bridge of SIP/craig-b8d1 and SIP/pstn-spa3k-5d4b

For some reason you are not advancing to the next stage. However, you seem to be authenticating OK, since you don't get a reject message from the SPA3000 when you try to dial (I misconfigured mine to see what asterisk sees). I wonder if it's a dialplan issue. What is your configuration on the PSTN Line tab for the VoIP-To-PSTN Gateway Setup?

As a secondary note, the firmware version you list is a bit old. You might want to upgrade to the lastest 2.0.13g.

[RMHunter]

Ahh.. that must be it...
Under "VoIP-To-PSTN Gateway Setup"

-- Gateway Enable:yes
-- VoIP Auth Method:HTTP Digest
-- PIN Max retry:3
-- One Stage Dialing:yes

-- VoIP Caller Default DP:none

further, all the dial plans are set "(xx.)" save DP8, which is "(<S0:1000>)"

What should I have there?