Plz Help :\ for zoom ata ?

Xtr
Posts: 43
Member Since:
2006-08-25

Hi every body. i need little help with my zoom ata 5801. (sorry for my english isnt very good) I need to connect my zoom for receiving call from pstn and route it to trixbox then i will pick up call with my ip phone.... and i need the same thing in reverse. Take my ip phone call with trixbox using my zoom ata conected to my home pstn line. Can someone already doed this ? plz is very important for me.

Any help will be greatly apreciated. You can write me to xtr(at)xtr(dot)cc

Ps: trixbox pro or Se no importance for me. if one work go for it :) (my big problem is routing incoming pstn call from my zoom to my trixbox and routing outgoing call from my trixbox to my zoom ata thats connected to my pstn line)

(i already read a lot of documentation but still dont wotk :( )



bubbapcguy
Posts: 3774
Member Since:
2006-06-02
That device

Hi, I do not think that model device supports that, the 5801 is a service provider model, not a end user device. (you dail 8 for VOIP and 9 for PSTN)

I think you need the Linksys 3102.



trevisa
Posts: 51
Member Since:
2007-06-19
provider model x end user model?

Bubba,
what is the difference between one kind or the other?
I just bought the zoom 5801 to do this bridging, and it has instructions on voip-info (http://www.voip-info.org/wiki/view/Zoom+5801), and on manufacturers site (http://www.zoom.com/documentation/voip/ATA_TechnicalReference.pdf), explaining how to do so.

I can make the voip to pstn bridge, but not the opposite way, and I dot no mean to do both at the same time.

If, I pick one of my softphones under asterisk and call the extension atributted to the ATA, it picks up and give the line signal, and so I can dial anynumber trough PSTN, but...

... if I get a PSTN phone and call my ATA number, it never picks up the call. Rings forever.
I´ve already tried to change techinical specifications thinking it might make a difference, but it does not so far.
If I pick up the phone, I get the line signal as well.

Thanks for any help.



Navigator2004
Posts: 2
Member Since:
2008-06-25
Did you get 5801 working?

Trevisa, did you get the PSTN to VOIP bridging working? I'm having the same problem here.

Thanks in advance.



trevisa
Posts: 51
Member Since:
2007-06-19
The 5801 is a pain in the ass...

Navigator,

I came to conclusion that this ATA is a pain in the ass, when connecting the PSTN to VOIP.
The FXS port works very well, good sound and everything.
But the FXO is trouble.

The only way I´ve managed to connect it to the asterisk was through the hot dial feature. Instead of the 7777, I used the number for one queue I´ve created.
But this option disable the normal use of the FXS port, as when you pick up the phone it dials to the extension number programmed on the hot dial.
And, I experienced some problem with the hang up tone, so a call that came in the PSTN would take more than a minute to hang up, after the remote party disconnected.
More, the ATA would only answer to the calls when rang direct from the operator, when I connected it to a AT&T Partner Plus PBX, the ATA would ring forever and never answers. Yes, I´ve tried every adjust possible on the ATA, ring Impedance, timers and so on.

It is impossible to use both ports at the same time also. And I have all the latest firmwares.
I must be very lucky, because the Zoom support team actually did talk to me, and remotely upgraded my firmware. Most of my searches on the internet says that the support is unresponsive.

It seems that the 3102 is a much better option, but I´m kind of traumatized with these ATA. Too much tweaking to make them work fine.
I´d only use the ATA in order to deploy FXS ports, nowadays.

Anyway, if you want I can try to take screen captures and send them to you.
Just for information, I´m in Brazil and I don´t know about regional settings too much. Maybe I´ve missed something because of it.

My Zoom today is put aside here, and has a SELL sign on it. Anyone interested?

Regards,



trevisa
Posts: 51
Member Since:
2007-06-19
Two stage dialing

Forgot to say that when calling from VOIP to PSTN, I´ve could only perform it with multi stage dialing. It means I have to call the ATA´s extension, it would ring, answer automatically(I´ve had programmed the ATA to answer automaticaly every call), and then it gave me the dialing tone from the PSTN.

This is not good, also.



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