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jslh

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Hi,

Is it possible to update the supported devices in Cisco Linksys as quite a number of the models has been EOL / No Longer Selling. Such as;

PAP2T, SPA 901, SPA 921, SPA 922, SPA 941, SPA 942, SPA 962, SPA 400

PAP2T has been replaced by SPA 112
SPA 400 has been replaced by SPA 8800

Does the replacement models like SPA 8800 and SPA 112 works with Kerio Operator?

On top it, we needs a gateway with 8 FXO ports, any recommended products that work well with Kerio Operator?

Regards
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Vladimir Toncar (Kerio)

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Hi,

SPA112 and SPA122 can be auto-provisioned from Operator since version 2.1.0 Beta 3. We also support a number of the newer SPA desktop models.

We currently have no recommendation for a 8-port FXO gateway, however we are going to test a Grandstream model soon.

Vladimir
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jslh

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Hi,

How about Digium TDM800 8 Ports Analog Telephony Cards? Since Kerio Operator supporting TDM410 which is a 4 Ports Analog Telephony Card.

Both are the same except that one is 4 ports whereas the other one is 8 ports.

Regards.
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Vladimir Toncar (Kerio)

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Hi,

TDM800 is currently not supported.

Vladimir
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uri

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Hello Vladimir!

Almost a year ago you have posted the message that you'll test grandstream fxo gateways!
And I have a question: did you'd finish your tests successfully? will some next releases contain provision for granstream gxw410x(for now it's FXO gateways)? for example 3CX do it without any problems!

Thank you in advance,
Alexey
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Vladimir Toncar (Kerio)

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Hi,

We have a knowledge base article that describes the configuration of the GXW 42XX gateways.

We currently do not support auto-provisioning for FXO gateways. Only some FXS gateways are supported, like the Cisco SPA8000.

Vladimir
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uri

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Hello Vladimir,

GXW42XX it's a FXS gateway and that's why the KB that you mentioned is not relevant for me, I'm interesting in GXW410x - FXO gateway. And also after firmware update web gui completely different! Sad

Will you test FXO Grandstream gateways?

And this is perfect that Operator starts(as I remember form version 2.1) support auto-provisioning for SPA8000 because I'm using it. Smile

Kind regards,
Alexey
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obtim

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+1 to uri
In the 3CX Phone System has long realized this possibility 3cx.com.ua/autoprovisioning-fxo-gateways-grandstream-gxw4104 -gxw4108-3cx-phone-system/
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uri

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Hello Vladimir,

Any update? will you test FXO Grandstream gateways and make a KB/auto-provisioning? or you don't have it even in plans?

regards,
Alexey
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Vladimir Toncar (Kerio)

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Not in the short-term plan as of now. Unlike the Grandstream phones, the gateway is not that easy to get, at least here in the Central Europe.

However we could provide some hints as to how to approach the configuration. For example, can the gateway behave as a SIP registrar?
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obtim

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Vladimir, I have this device. In a week I could join it to my system and give you syslog if you need.
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uri

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Hello Vladimir!

GXW410x can not behave as a SIP registrar(Russian grandstream support told me that).
Full description of the device : http://www.grandstream.com/index.php/products/ip-voice-telep hony/enterprise-analog-gateways/gxw410x
The main problem is incoming calls, it just don't route calls to Kerio operator, but in operator looks like registered lines! I've tryed to do like in this thread http:// forums.kerio.com/t/26155/anyone-using-a-grandstream-gxw4104- fxo-gateway-with-operator- but useless .
For outgoing calls, at last, I've make straight to gateway port routing through adding prefix to phone number(for example number 1234567890, by leading digits 1234 Kerio see that it is mobile phone number from operator A, and you know that on your gateway to port 5 connected GSM gateway with SIM card from operator A => in Kerio we add extra prefix 995 to number and call goes to port 5)

regards,
Alexey
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Vladimir Toncar (Kerio)

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Maybe you are trying to use the gateway as a regular internal extension. That's the approach for FXS gateways.

As to FXO gateways, it's better to create an external interface in Operator and use it without registration. I've used this method in the SPA8800 howto. The general approach is as follows:

* Create a SIP interface in Operator without registration.
* Disable registration on the SIP account in the gateway.
* Use the same SIP authentication ID and password on both sides (ie. the gateway and SIP interface in Operator)
* Since your gateway is not registered to Operator, you must set up some routing rule that will send all calls arriving from the analog line to Operator's IP address. We achieve this using a specially formatted dial plan field when working with the Cisco/Linksys SPA devices.
* Pay attention to having the phone numbers right in both the gateway and Operator. If the phone number of your analog line is "555123456", use this same number as the external number in the Operator SIP interface. Then, when sending the analog call from the gateway to Operator, you simply send it to "555123456@operator_ip"

Hope this helps.
Vladimir
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uri

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Hello, Vladimir!

I'd try to do like you say but useless. Could you help me to solve this problem?
And one more question about outgoing calls: line a with prefix 1. if the line busy and I try to call 1xxxxx the call is not happen. Can I change something to transfer call to another line in such situation?

regards,
Alexey
Vladimir Toncar (Kerio)

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Well, I can only guess that these calls are routed via the analog gateway. So the number of concurrent calls depends on how many analog lines you have and it also depends on whether the gateway can use the second (third) line when the first one is busy.

It is also possible to set the number of concurrent calls in Operator's external interface, so check what is your value there.
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