Need help configuring Zoom 5801 for Asterisk

[josetann]

Just got a Zoom 5801 yesterday. Having some trouble setting it up to work with Asterisk. First problem is that it came with firmware 1.2.5 but then downgraded itself to 1.2.4 (but still has config version 1.2.5). I tried live help on Zoom's site, there's never anyone available. Tried to email support, they replied this afternoon that they needed everything under the System Information screen (which I copy/pasted, but they already had my serial number and my model number). So it may be a while before I get back to 1.2.5.

Anyways, I cannot get the 5801 to send CID info to Asterisk. I'm calling my home number which is connected to the FXO port of the 5801. It passes it off successfully to Asterisk, but Asterisk just uses the extension number assigned to the 5801 (this is also the username programed into the 5801, in case that's how it's getting it). I believe I have everything configured correctly, "Bridge From PSTN to VOIP", "Auto-Answer PSTN (FXO) Calls", "FXO PORT Accept Any Call", and "FXO Port Accept Anonymous Call" are all set to Enable, while "Caller Password" is Disabled under the "Bridging From PSTN to VoIP" screen. Under "Misc FXO parameters" I have "FXO Port Display Caller ID" and "FXO Port Caller ID Sent After One Ring" both set to Enable. Other settings under this screen are:

FXO CID Wait Duration (10 ms) - 500
FXO CID Clear Duration (10 ms) - 1000
Billing Delay Duration (10 ms) - 100
FXO Security Entry Duration (10 ms) - 1000
Non-VoIP Calls (Click on Help for details - blank (it's actually blank, and btw clicking on help shows that there's no help for this page)
Auto-Add Area Code when Dialing non-VoIP local calls - blank
Number of Digits dialed for local calls - 6

I'd also appreciate if someone can post an example of calling out through the FXO port using Asterisk, with single-stage dialing. Right now when I dial any number the 5801 simply gives me a dial-tone. I believe single-stage dialing wasn't added until 1.2.5 (which I now don't have), but I'd like to be prepared once they finally get me back to 1.2.5.

[josetann]

Took a while, but I was able to get back to 1.2.5 on my own. One-stage dialing and CID are now working just fine. I do have a bit of a problem with the 5801 not knowing when the line has disconnected (i.e. when a call comes through the FXO port, it won't hang up on its own, asterisk has to time-out and hang up itself). I can live with that for now though.

[Navigator2004]

Could you please tell me how you reverted back to 1.2.5. I'm having the same problem.

Also did you get your PSTN to VoIP Bridging to work? I get the Dial Tone, but can't get Bridging to Call out on my VoiP.

Thanks

[josiegross]

I was just looking at this post and was wondering if in fact you have been able to pass CID from the PSTN to the VOIP side.

I am looking to do this so that a call comes in from the PSTN and it gets passed to Asterisk with CID but all I get is the extension that the 5801 is configured on. I have the 5801 configured as a custom trunk but was wondering how you have it.

All my other settings are consistent to yours.

Thanks