[jhguan]

How I can unconditionally send local calls out via PSTN ?
--Hong

[RareSanity]

Hello All,

I have a quick question related to this hw/sw combination. Is it possible after getting everything setup that you can make use of the PSTN call waiting? From my experimentation it seems like there is no way to have the "flash" get passed through the Asterisk server to the SPA3000.

So when I am talking on the phone connected to the SPA3000 through the PTSN line and I get a call waiting alert, (from the PSTN line) if I press the flash button on the phone, I get a dial tone (to start a 3way call) produced by the Asterisk box as opposed to "answering" the incoming call.

Is there anyway to alleviate this problem? Any insight would be appreciated.

[sethg]

Could anyone who has used the SPA3000 as an Asterisk FXS/FXO adapter as well as conventional PCI cards comment on the pro's and con's of implementing a small Asterisk system with either type of hardware? The SPA-3000 has an obvious advantage in that when power fails, it automatically connects FXS to FXO, so you have emergency phone service. I don't know if any of the PCI FXS/FXO cards do this when power goes away or the server is dead. I also wonder about differences in the quality of echo cancellation and the use of the minimum length jitter buffer needed for a given connection. The latter is important to reduce latency. My suspicion is that the PCI cards have the advantage there, but only if the filtering is done by on-board DSP's, not by the the CPU in the Asterisk box.

I'm interested in learning all I can about this.

[sydwahl]

Hi all got a new SPA3102 not able to get into configuration page through web browser..keeps timing out and not able get in to set up for use. Any suggestion out there before I consider sending it back.

[tesluk]

Originally Posted by sydwahl

Hi all got a new SPA3102 not able to get into configuration page through web browser..keeps timing out and not able get in to set up for use. Any suggestion out there before I consider sending it back.

sydwahl, I believe the SPA3102 has its web-based interface dissabled disabled by default. Use the IVR (interactive voice response) menu to enable the web-based utility interface.
**** enters menu
7932 # enters web-based utility setup
1 enables the web-based utility (0 disables it)
Don't feel badly - this new 'feature' will be causing many people frustration and RMA's until it is communicated better to users.

[jchuby]

come someone post how to do this wiht AMP.... i do not want to start editing config files since AMP has been doing my settings all along.

I used the Voxilla Wizard to set up my SPA-3K - the two extensions have programed into my asterisk box. 201 for line 1, and 200 for pstn incoming.

SO if someone can post what to do now in AMP or FreePBX so that:

1) Incoming calls over PSTN still ring Line 1 (i had ring through enabled before because of a fax machine on that line)
2) Outoging calls can be made of SPA-3K PSTN (For emergency # use)

3) If someone could tell me how to have the incoming PSTN calls be checked for a fax that would probably help make my life easier.

**i am an asterisk newbie, using Trixbox with FreePBX.
My extensions and trunks are setup and working, now id like to get the SPA-3k working

[toddvelie]

Originally Posted by jhguan

How I can unconditionally send local calls out via PSTN ?
--Hong

Unplug the device.

Actually - It all has to do with your dialplan. You need to find a numeric pattern or set of patterns that will catch all of your local calls. In the US, some people are lucky enough to have their entire area code be considered local, then it's easy. (But, I'm not one of them.)

What's unique about local calls to you? Do they have a different number of digits? Do they start with a certain set of prefixes? Once you post that, someone should be able to help you.

[Inept]

Is it possible to get the sipura to answer all the incoming calls and forward them to the asterisk server after x amount of rings. I have setup a few voicemail boxes and would like the caller to be able to leave messages on them. What info would I need in my sip.conf and extensions.conf?

[mberlant]

inept,

Your other thread talks about this situation directly. Kindly return there to continue your quest to get your SPA3000 working with your Asterisk.

Before you can run (send inbound PSTN calls to an IVR for capture into various mailboxes) you need to learn to walk (implement an "s" stanza in your extensions.conf context).

[bernart]

Can multiple asterisks "dial" out through a single SPA Voip-gateway or must it be the only one that the PSTN line is registered to.. the PSTN line (voip gateway) is registered to a specific server but can other servers that the SPA is not registered to route calls thru the VOIP-gatewy?

can any server do: exten => _#9.,1,Dial(SIP/${EXTEN:2}@45.76.98.87:5061,60,tr) even if the spa is not registered to them.