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

[admin]

Hello Dan,

have set up as suggested but..nothing...still the same.

What can I do next?

[Dan Lukes]

We still don;t know the Vohippo requirements, thus very hard to suggest something valuable. Try:

DTMF Tx Method: AVT
DTMF Tx Mode: Strict
DTMF TX Strict Hold Off time: 80 (or even more)

[admin]

Hello Dan,

sorry I was wrong! I had not updated configuration that's why wasn't working. But still have part of the problem. Now when digiting #1 the voice says: TRANSFERRING and now as Vohippo support says I should include the number of the internal phone but...nothing happens...

I contacted them again and they say that after the voice I hear I should only digit the internal no. (201)

Any other suggestion?

Maybe we are almost there??

Thanks very much in advance for your help!

Armando

[Dan Lukes]

So DTMF sequence #1 is recognized. We can assume DTMF is recognized properly now at all. Thus even DTMF sequence 201 is recognized.

So sorry, but SPA112 is no longer issue cause - signals are delivered per Vohippo requirements (assuming the 201 is the valid internal line number recognized by Vohippo).

There's litlle I can advise you - only Vohippo can disclose why transfer has not occurred despite properly requested.

But message "TRANSFERRING" sounds suspicious when used BEFORE you entered the target number. Moreover, I tried to read documentation on Vohippo website. It's very short and there's even no single letter dedicated to transfer. Also, I see so many links no nonexistent pages. It doesn't look so serious. I'm unsure they know what they are speaking of ...

[admin]

Hello Dan,

it's working. my mistake was in hanging up too early. Basically need too wait almost 15 seconds for the call to be transferred. How can i reduce such time to almost immediately?

Awaiting yours,

Armando

[Dan Lukes]

This method of transfer is out of our control - all procedure steps are handled by remote switch.

I can just guess what's causing such delay on remote switch.

I assume it's waiting for complete number entered or timeout. Just 2001 is not recognized to be complete number, thus, it's waiting for timeout - and it may be 15 seconds.

You can try to dial 201# instead of just 201 - the # is often recognized as "dialing completed" character.  Or there may be a longer, canonical form of 201 short number. But I can't guess such form.

But there's other explanation possible - the "short transfer time" is paid feature. You can either wait or pay for no-wait service.

Sorry, only Vohippo can disclose their internals ...

[admin]

WORKING WITH #1201#

YOU ARE JUST A GENIOUS

THANKS INDEED

ARMANDO

[Dan Lukes]

Just a lot of experience.

Glad to hear you solved it.

[Dan Lukes]

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

So no two external lines, but the single one. Hm ...

There's feature called Synchronized Ring - both internal phones should ring for incoming call when enabled. Unfortunately, such feature may not work as expected (read: spa112 Synchronized Ring not working). But even if it will work, you still can't transfer calls from line to line. SPA112 can't do something like it by self.

Thus the external PBX is necessary here. But single SPA112 with proper configuration should be enough.

By the way can you suggest any other virtual pbx software which easier to use?

So sorry, I'm on other end of barricade. I'm using plain Asterisk for everything. But Asterisk is rather building it than complete solution. It require a lot of skills to be used (but then you can do almost anything you can imagine).

So Asterisk is everything but "software easier to use". Sorry, I'm familiar with no plug&play SIP soft switch ...