[muppetmaster]

Posted this on another site, though it might be useful here if not already embedded somewhere in a thread:

Sipura 3000 -> PSTN Config for Incoming

- Add a dialplan (ie - Dial Plan 1/Dial Plan 2) with a SIP URL destination as follows:

S0<:1000@192.168.1.50> (where the IP address is that of your * instance, also ensure the extension is properly accessible in you extensions.conf)

- Enable both the VoIP to PSTN Gateway and the PSTN to VoIP Gateway

- Set the default dialplan under PSTN to VoIP Gateway to the dialplan where you inserted the above SIP URL

Asterisk Configuration for Outgoing

- extensions.conf

[globals]
PSTN_GW=192.168.1.51:5062

[pstn]
exten => s,1,Dial(SIP/${EXTEN}@${PSTN_GW})

If behind a firewall this should be okay. If not, then of course you would want to configure with HTTP digest in mind.

[PhoneBoy]

Worth a sticky.

You might want to configure with HTTP Digest in mind anyway. Here's what my configuration looks like (note I had to pull a nightly * from CVS to make this work right):

PSTN Tab:

SIP Credentials (Proxy, User ID, Password) point to an extension in sip.conf. This allows the SPA3000 to make a call to the * box as well as allow the * box to call the SPA-3000.

Dial Plan 8: S0<:666> (666 is an extension on my * box)

VoIP-To-PSTN Gateway Enable: Yes
VoIP Caller Auth Method: Yes
One Stage Dialing: Yes
VoIP User 1 Auth ID: user
VoIP User 1 Password: password
VoIP User 1 DP: none (this means pass whatever the remote end sends)
PSTN-To-VoIP Gateway Enable: Yes
PSTN Caller Auth Method: None
PSTN CID For VoIP CID: Yes
PSTN Caller Default DP: 8
PSTN Answer Delay: 5
VoIP Answer Delay: 1

In sip.conf, I have my SPA3000 PSTN Line extension (54) defined as follows (I also have a seperate extension for "Line 1" of the SPA-3000 with identical settings):

[54]
; spa3k line 2 (pstn)
type=friend
host=dynamic
context=home
secret=whatever
callerid="PB SPA3k PSTN" <54>
mailbox=54
dtmfmode=rfc2833
canreinvite=no
nat=0

Also in sip.conf, I also have an entry to make outgoing calls via the SPA3000:

[pstn-spa3k]
type=peer
auth=md5
secret=areyououtofyourmind
username=pinky
host=spa3k.phoneboy.com
fromuser=splat
port=5061
dtmfmode=rfc2833
nat=no
context=home

In extensions.conf I have:

exten => _#9.,1,Dial(SIP/${EXTEN:2}@pstn-spa3k,60,tr)
exten => _#9.,2,Playback(abandon-all-hope)
exten => _#9.,3,Congestion.

Basically, this setup allows me to route my PSTN Line (which is actually a Broadvox-supplied SPA-2000) into my Asterisk server. I can also make calls out this device from other extensions on my Asterisk server.

[muppetmaster]

I did some testing. And even though I share the same config as you above (on the 3k and on Asterisk) I can still dial in both these ways via the PSTN GW:

exten => Dial(SIP/${EXTEN}@192.168.1.10:5062)

or

exten => Dial(SIP/${EXTEN}@pstn-spa3k)

Shouldn't the first one fail as it is not authenticating as the second one is?

[PhoneBoy]

I believe the first one should fail.

[pupfuzz9]

Originally Posted by muppetmaster

I did some testing. And even though I share the same config as you above (on the 3k and on Asterisk) I can still dial in both these ways via the PSTN GW:

exten => Dial(SIP/${EXTEN}@192.168.1.10:5062)

or

exten => Dial(SIP/${EXTEN}@pstn-spa3k)

Shouldn't the first one fail as it is not authenticating as the second one is?

If the first one works, then either you don't have HTTP Auth enabled in your SPA3k, or your asterisk box's IP address is listed in the VoIP access list field?

Another thing to test is to use the first line, and then remove the configuration for your spa3k from the sip.con file. Then you'll know for sure.

[simonb]

Is there any way to get the SIP extension to ring at the same time as the PSTN is still ringing, all I can seem to do is either increase the delay before the SIP call is made or if I reduce it the SPA answers the PSTN call then calls the * SIP extension.

So what I want to do is : PSTN rings, SPA3K calls an * extension via SIP, asterisk then dials other extensions in a hunting group, one of the asterisk extensions answers, then SPA3K answers PSTN.

[muppetmaster]

Originally Posted by pupfuzz9

If the first one works, then either you don't have HTTP Auth enabled in your SPA3k, or your asterisk box's IP address is listed in the VOIP access list field?

Another thing to test is to use the first line, and then remove the configuration for your spa3k from the sip.con file. Then you'll know for sure.

HTTP Digest enabled (otherwise wouldn't the second one fail?). Also, I don't recall adding antying to a Sipura Access List. Where is this done?

[pupfuzz9]

Originally Posted by muppetmaster

HTTP Digest enabled (otherwise wouldn't the second one fail?). Also, I don't recall adding antying to a Sipura Access List. Where is this done?

I was suggesting that you remove the MD5 stuff in asterisk's sip.conf file. Then you will know if it is using HTTP Digest or not.

As for access list -- that is a field in the PSTN tab of the SPA3000

[bernart]

VoIP Caller Auth Method: Yes

given the tone of the note, and that there is no "yes" choice. should this have been:

VoIP Caller Auth Method: HTTP Digest

[bernart]

A few questions about interfacing the SPA3k to asterisk.

1. Can line 1 be configured to interface with the * box so that I can place and rcv calls via the FXS port. Meaning that I can ring the analog phone attached to the port or use the analog phone to place a call thru the * box

2. Can the asterisk box support both clients (Line 1 and PSTN) simultaneously, given that both clients share the same IP address except for the port number

To date I have created an extension in SIP.conf representing Line 1 client simillar to Phoneboy's [54] and defined the Line 1 SIP credentials (proxy, user id, password) to reference the extension but I can't get it to communicate with the * box. The spa3k "info" page shows it as registered but I cannot ring the extension or cause it to "trigger" the context in the extensions.conf. So is my objective doable and if so what am I missing