Cisco SPA 112

[admin]

Hello everyone,

it's my first post and as generally happens is a a request post for help with the product in object. Basically I have installed it and set up with the provider messagenet and a analogic Panafonic phone. Everything runs perfect with one phone and also with the second one but I do not not how can I transfer calls between phone (provided same is possible). Could anyone help me out please?

If same is not possible, can I use a virtual pbx installed on a computer for making internal calls or transfer a call internally still using the Cisca SPA 112?

Thanks in advance for your help,

Kind Regards

Armando Scotto

[Dan Lukes · Accepted solution]

SPA112 is just SIP<->POTS bridge. It lacks "PBX intelligence". You can transfer call from line to line as long as it's supported by PBX the SPA112 is registered to. Exact method how to invoke such feature vary from PBX to PBX, so follow switch documentation.

can I use a virtual pbx installed on a computer for making internal calls or transfer a call internally still using the Cisca SPA 112

SPA112 will register to any SIP softswitch as long as there's IP connectivity available. It doesn't care (in the fact it doesn't know) the switch is virtual or not. Regardless of the exact meaning of the term 'virtual' you used here.

I can neither confirm nor reject assumptions related to softswitch capabilities (you mentioned making call, call transfer, ...) . It's matter of softswitch you decide to use and it's configuration. It has nothing to do with SPA112. As I mentioned above, SPA112 is SIP to analog bridge only.

[admin]

Hello Dan,

thanks for your help. Despite I can use the computer well and I am autodidact in many things like setting up the Cisco ATA, I am not an IT therefore sorry for my simple writing and my english too. Now I have two more question for you:

first is if I install a virtual pbx on the computer, shall I still use 1 Cisco SPA 112 for both phone for which I want to transfer calls or I need two indipendent ATA?

Yesterday I tryed to install Axon Virtual Pbx but setting up the data provider, extentions etc on the Pbx but now what shall I put on the provider data of the ATA?

Thanks

Armando

[Dan Lukes]

if I install a virtual pbx on the computer, shall I still use 1 Cisco SPA 112 for both phone for which I want to transfer calls or I need two indipendent ATA?

I see no reason for two ATAs. But, may be, I missed the matter or a feature you wish to have. You didn't described the overall goal of the project.

Why you thing the second ATA may be necessary here ?

I tryed to install Axon Virtual Pbx but setting up the data provider, extentions etc on the Pbx but now what shall I put on the provider data of the ATA?

I'm unsure I understand the question. Moreover, I'm not familiar with Axon's software.

It seems you have two independent VoIP lines from your provider - two numbers, two pair of name and password. Those needs to be configured into Axon as external lines. Sorry, I don't know how to do it on Axon.

Then, your SPA112 needs to be introduced to Axon as two internal SIP phones (internal extension). Once more, sorry, I don't know how to configure it on Axon.

Please ask for help in Axon's forum.Note the guide for SPA3102 on top. Whole SPA3102 use different GUI that SPA112, the basic principles of configuration are same. Thus the guide for SPA3102 may help you to configure SPA112.

[admin]

Hello Dan,

the overall goal for the project is to have 1 external line which can be picked up by both phones (I am here now) and calls can be transfered between the internal extension.

Now since same can not be done only using Cisco SPA 112, I need to configure Axon with the data I configured Cisco before and than I need to configure the ATA with the extensions but I couldn't find any help on Cisco website or on the net.

I will follow your advice and check in Axon's forum the guide to SPA3102. By the way can you suggest any other virtual pbx software which easier to use?

Thanks

Armando

I am presently stucked at Cisco configuration when configuring Axon

[admin]

Hello Dan,

I just check but it seems that it talks of line 1 and pstn line. I need a configuration of Line 1 and Line 2 for me a my colleague and each of us can take the call when the phones ring and each of us can pass the call to another one.

Thanks a lot

Armando

[Dan Lukes]

So you need to create a group, both lines needs to be member of such group and incoming call needs to be directed to such group.

So sorry, just guessing - I have no experience with Axon. The paragraph above describes just principles using generic terms. Axon documentation may use other name for the same feature. In the worst case, required feature (group call) may not be available in free mode and particular paid license may be required.

Ask Axon users for help ...

[admin]

Hello Dan,

hope you are well. Thanks for all your informations/suggestions of last month.

I have finally set up a new provider (Vohippo) defenitely better for quality of calls and tariffs and I have set up their own virtual switchboard which working pretty well but unfo I can not manage to transfer calls between the two internals.

Their code for transferring to an internal is #1 and they say that should work like this however is not and they asked me to check with CISCO...

Do you have any idea on how I can make this work?

For your info the ATA SPA 112 is set up as Factory defaults, only sip provide, user and password for the two lines were changed.

Awaiting your comments

Regards

Armando

[admin]

Hello Dan,

wanted also to mention that the DIAL PLAN as Factory default on SPA 112 is:

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Regards

Armando

[Dan Lukes]

Factory default configurations of SPA* are US-centric. It doesn't fit non-US environment well.

[admin]

What you mean sorry?

How can I solve my problem?

Regards

Armando

[Dan Lukes]

You claimed

wanted also to mention that the DIAL PLAN as Factory default on SPA 112 is: ...

It's not declaration of problem of any kind, it's just notice.

Thus I just explained why the factory default dial plan look like this one.

[Dan Lukes · Accepted solution]

Their code for transferring to an internal is #1

Unless I missed something, Vohippo ask you to send DTMF signals #1 to initiate transfer. In such case, the format of DTMF transmission become very important.

DTMF can be passed thru SIP

1. in-band (as true sound mixed into audio stream)
2. as NTP (specially marked event payloads in the RTP stream) (also know as RFC4733)
3. as SIP NOTIFY message
4. as SIP INFO message

Which one method is recognized by Vohippo ? Have you configured supported method on your SPA112 ?

[admin]

Thanks,

sorry for my very low IT but as far as I understand they told me that digiting #1 I should hear a voice which will ask to digit the internal desired (201 for example which is for my user that I want to transfer the call to).

I haven't configured the supported method, I have left everything as factory default because I don't know how to set up. I will attached a coupld of pictures to show you the VOICE settings for my line.

Regards

Armando

[Dan Lukes]

But we know no Vohippo requirements yet so we are unable to follow them.

Well, I can try to guess ...

Configure G711a as "Preferred codec", second and third preferred codec set to "Unspecified". Set RTP Packet Size to 0.020

Try again.