[Asterisk-Users] Sipura 3000 Analog Line No Answer, No Audio

[Tim P]

Problem 1 - Outgoing:
I am able to call out of the * box using the analog line attached to
the sipura 3000 but when the person being called answers there is no
audio from either end.  * registers that the call was answered but
passes no audio.

Problem 2 - Incoming:
When calling into the 3000 attached to * it never seems to pickup the
line.  The phones don't ring on the asterisk side.

I used the below writeup to create the extensions for Line 1 and PSTN
in the 3000 as well as creating the Trunk, extensions and DID routes
in *.

Can someone give me an idea of where to start with the troubleshooting
here?  I am kind of lost as to where to begin.  Thanks!

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In AMP add an extension (e.g. 200) to correspond to Line 1 on the SPA,
ensure that port is 5060 and context is from-internal. Add a second
extension (e.g. 280) for PSTN Line on SPA, ensure that port is 5061
and set context to from-pstn (disable voicemail & directory on this
extension).

In Trunks add a Sip trunk and copy the Outgoing block as follows (just
leave Incoming as it is - do not delete the any defaults, but you do
not need to change them either).:

Trunk name sipura1 

context=from-pstn 
fromuser=280 (or whatever extension you used) 
host=IP address of you SPA (needs to be fixed IP) 
port=5061 
secret=your password 
type=peer 
username=280 (or whatever extension you used) 

Inbound User context sipura1-in 

Leave defaults in Inbound box and leave Register String blank. 

In DID Routes, add DID with a unique string (I used S followed by the
PSTN number that the SPA is attached to - e.g. S12345678

Set an outbound route using the new sipura1 trunk. 

On the SPA 3000: 
Do the following configuration in admin login, advanced mode: 
In Line 1, make sure SIP port is 5060, & proxy points to your * Box,
no outbound proxy. Fill out subscriber info with settings above e.g.
User ID = 200, Password =***, Display Name =***. Set your preffered
codec.

In PSTN Line, ensure SIP Port = 5061 & proxy = Asterisk Box IP, no
outbound proxy. Fill out subscriber info with Display Name =****, User
ID = 280 (or whatever you used), & Password =****. Set preferred
codec. It is vital that you Set Dial Plan 8 to (S0<:S12345678>) (or
whatever string you used for the DID route in Asterisk).

Ensure that both VoIP-To-PSTN Gateway Enable and PSTN-To-VoIP Gateway
Enable are set to yes.
Set PSTN Caller Default DP to 8. 
If you want incoming calls to all be sent to * then set PSTN Ring Thru
Line 1 to no.
Set PSTN Answer Delay to the number of seconds that you want the phone
to ring for before sending it to your * box.

Leave other settings on the SPA at factory defaults until you really
know what you're doing and want to fine-tune things.

Lastly, make sure you plug into the line jack into the SPA and not the
jack marked phone! I know this seems obvious, but I've missed this
simple step before!

The only kink with inbound using the settings posted is that you can't
have it ring to a phone plugged into the Sipura's phone port. You can
still call out, and the system will still pick up the call if you have
auto attendant recieve the calls. But, if you set the inbound calls to
ring extension 200, your calls will just go directly to voicemail.

That aside, you can have any other phone on the system ring for
inbound calls directly, or set a ring group.
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