Connecting Asterisk to legacy PBX systems?

I realise Asterisk can be conected to a legacy PBX system using FXO/FXS analogue cards. Is anyone doing this over a digital connection ?

I’m puzzled as to why VOIP hardware providers are producing analogue gateways and products and not digital ?.

If I understand this correctly, an analogue connection between say an Asterisk server and existing PBX is limited to one simultaneous call, but digital can support more.

Is anyone using digital ? how have you done it ?

thanks, :wink:

Hi there,
Yes I have put an * box between a telco (BT) and an exsiting PBX (Samsung DCS). After the first couple of problems all works fine.
I used a quad span PRI ISDN card (30 concurrent channels). Span 1 went to Telco and the 2nd span went to the DCS.
I think that the main problem was the issue of timing source.

Regards

Jonathan

[quote=“ac931274”]Hi there,
Yes I have put an * box between a telco (BT) and an exsiting PBX (Samsung DCS). After the first couple of problems all works fine.
I used a quad span PRI ISDN card (30 concurrent channels). Span 1 went to Telco and the 2nd span went to the DCS.
I think that the main problem was the issue of timing source.
[/quote]

Has anybody done it with Ericsson MD110?

Probably I’ll have to connect Ericsson MD110 with Asterisk but I’m not sure what to expect. I would presheate any info or hint.

Tomislav

Jonathan,

Thanks for your comments.
I would like to use Asterisk as an addition to a customers existing PBX as a VOIP gateway to say other sites, but keeping the solution generic as possible without too many alterations on the customers existing environment.

Ideally I would like to unplug one of the customers existing digital phones and plug it into an Asterix server, then using that extension to route the VOIP calls. Am I talking rubbish here, is this sort of approach possible ?

thanks, :stuck_out_tongue:

[quote=“tangerine007”]Jonathan,

Thanks for your comments.
I would like to use Asterisk as an addition to a customers existing PBX as a VOIP gateway to say other sites, but keeping the solution generic as possible without too many alterations on the customers existing environment.

Ideally I would like to unplug one of the customers existing digital phones and plug it into an Asterix server, then using that extension to route the VOIP calls. Am I talking rubbish here, is this sort of approach possible ?

thanks, :stuck_out_tongue:[/quote]
This is a Perfect application for interconnecting 2 PBX’s that are not VOIP Compliant.

Asterisk Box A and B have a PRI Cards

Pri on Each Asterisk is connected to the PBX at the location it resides.

Program the Pri’s to work with the PBX as DID OR Tie Line Arrangement Trunks.

Next Program IAX Trunks Between the Asterisk Boxes and allow as many channels as your bandwidth between the sites allow.

Once a connection between each asterisk is done and the pri’s are up , program routing information to route calls from the pri inbound > out the iax > to the inbound iax > to the outbound pri > to the remote pbx and vice versa.

Kinda quick and dirty explanation but you get my drift.

I did this with 2 Asteirsk Boxes and 2 Nortel PBX’s

[quote=“plasmaflow”]
I did this with 2 Asteirsk Boxes and 2 Nortel PBX’s[/quote]

Nice. Can you tell me, those two Nortel BPX’s, are they digital? Did you have any experiance with Ericsson MD110?

Tomislav

plasmaflow,

Thanks for the explanation, how many simulataneous calls can the pri from your Asterisk to PBX cope with ?

Does a digital phone use a PRI port ? Or do I need to be looking for something else on the PBX.

thanks,

I guess that depends on what you mean by a digital phone.

  1. I have not heard of a digital phone using PRI – that is a connection used for PBXen, not for individual phones.

  2. One meaning of “digital phone” with which I am familiar is an ISDN BRI phone, and that can talk to Asterisk with an appropriate adapter. There are PCI boards from Beronet and Junghanns – see the Asterisk Wiki for more info. There are also Ethernet-to-BRI gateways, for the cheaper version look at the offerings from AVM/Fritz! (a German manufacturer), for the more expensive version look at the SmartNode line of products from Patton Electronics/Inalp (Swiss-developed technology now manufactured and distributed by a US company).

  3. Another meaning of “digital phone” would be the various proprietary “communications terminals” offered by various PBX makers; according the the Asterisk Wiki there are ATA-equivalents for a couple of makes of these (if I remember correctly, Nortel Meridian is one of them), but for the most part they are just that – proprietary – and cannot be used with anything other than the PBX they came with.

Everyone,
Which (digium?) card is being used to connect legacy PBX’s to Asterisk?

[quote=“presidentscroob”]Everyone,
Which (digium?) card is being used to connect legacy PBX’s to Asterisk?[/quote]

Yes, on the trunk side by using a tie-line between the PBX and the Digium card in use.

And the model number of the digium card is…?
and is this the same card being used to connect to the pstn?
Basically, Im trying to have asterisk sit between our panasonic pbx and the PSTN and route calls depending on the number dialled (over the internet or or PSTN). Im trying to find out what digium card or cards I need to achieve this.
The line we have is a PRI ISDN line, we have 8 lines.
Any help Greatfully appreciated.

This would be the Digium TExxx series, the numbers indicate such things as the PCI bus voltage, number of T1/E1 ports on the card, and the presence of special echo cancellation circuitry. Look at www.digium.com/index.php?menu=product_c … y=hardware
for more info.

Alternatives include the Sangoma A10x cards, where x indicates the number of ports on the card (1, 2, or 4).

All of these provide 30 voice channels per port in E1 mode, or 24 channels in T1 mode; the channels can be divided between telephony and internet connectivity (provided your telco supports this).

for parcina: i’ve just completed integration of asterisk with Ericsson MD110, but i must patch either openh323 and asterisk source.
I connect MD and Asterisk via IP-Trunking. (Gw-to-Gw).

Obviously only basic calls works fine. Any services will work.

Best Regard