Gigabit Switch needed? SPA514G

[ncixbrian]

Hey guys,

I currently have SPA514G which has 4 lines connected to a PBX. I plan to have max 6 phones. But I need to have more lines, planning to go with SPA508 to have 8 lines, but then I will lose the Gigabit option.

I don't connect the phone to any PC port, useless option for me, I don't even know what it does.

Anyways, do I need the gigabit option? Is it worth it? Does it bring anymore more for my usage? Or 100mbps is good enough?

Thanks

[Dan Lukes]

Even SPA509 with 12 lines have just one handset. So only one audio stream at the time (there may be other calls with no audio path active). Audio codec supported require 64kbps or less.

Well, some overhead, some SIP signalling, ... 256kbps connection is more than necessary for all those VoIP thinks.

You claimed that you connect no PC to phone, so there is no other bandwidth consumer.

You have 100Mbps line for 256kbps of data.

Is it answer to your's question ?

[ncixbrian]

yep!! That means more than enough for even 6 calls?

Thanks!

[Dan Lukes]

Regardless of line keys you have, regardless of number of calls you have on hold, there is no way you can speak to 6 peers at the same time.

You can create 3-way conference, e.g. two peers at the same time. It is maximum you can - two audio paths at the same time.

[ncixbrian]

Then what is the purpose of gigabit switch associated to spa514G?

[Dan Lukes]

You may wish to chain a PC with phone and such PC demands more than 100Mbps.

But I don't recommend to chain PC with phone unless you can't avoid it.

[ncixbrian]

edit: ok

[Dan Lukes]

Note that this message is response to PARK SOFTKEY QUESTION question, not to the question above.

Don't ask why the response is here. That is it. Just take it.

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I can't seem to create a new discussion thread for this below: no idea why!!

Both of them has been placed on moderator's queue. They will not be published until approved.

 I tried twice..

Not exactly, you asked 3 questions in both postponed question, now you are asking 4.

Well, I will respond you here, but delete both superfluous thread once published, please.

Can I replace 192.168.1.111 by writing exactly the following: "$PROXY" ?

Yes, as long as proxy is set to 192.168.1.111. Did you tried it ?

1) PSK 1: fnc=sd;ext=70@192.168.1.111;vid=1;nme=Park
What is the*vid=1* do?

It mean that number 70 will be called in the context of Ext 1. If no vid parameter is present the first configured Ext is used instead.

Note that regardless of vid value or even vid parameter presence, the Speed Dial will initiate new call with no relationship to other calls (if any).

if we do not remove vid=1, only calls from located on this BLF button can be parked while other calls below this "line" can't be parked?

Parking is feature of PBX. Phone know nothing about it. The softkey you created will just call '70' with Ext 1 credentials. Further steps is up to PBX. There is no generic response to "can be particular call parked" question. It will be parked if PBX will decide to do so.

Extended Function:fnc=blf+sd+cp;sub=71@192.168.1.111;nme=P1

However, everytime I answer a call and accidentally pressed those lines keys, the call hangs up right after playing the asterisk message "I am sorry this is not a valid extension. Please try again". The message is played TO THE CALLER.

It is SpeedDial key. It will put current call on hold then it will initiate brand new call to number 71.

The message in question seems to be related to Asterisks misconfiguration related to hold call processing. Increase verbosity and debug level on Asterisk console and read the messages.

The phone has nothing to do with the issue you described.

I heard someone said "Setting SIP > "Keep Referee When REFER Failed" to Yes on the Cisco phone" fixed the problem, but it didn't for me. I obviously have no idea what this does.

"Keep Referee When REFER Failed" is related to call transfer. You do no transfer so it doesn't apply to your's case.

3) What is the difference between Share Call Appearance: Shared vs Share Call Appearance: Private

Does it has any impact if I use shared?

Did you tries to read Administrators guide ? It's valuable source of informations ...

The first one is for shared lines - e.g. for calls ringing on more than one phone at the same time. The second one is for lines that are not shared, e.g. it's only phone with particular number.

[ncixbrian]

you've been very helpful!

Yes, I did read some part of the admin guide, but I don't necessary understand what it does in real life.. I am very new to this, but your answers guided me to either confirm what I think is right.

When you said "It mean that number 70 will be called in the context of Ext 1. If no vid parameter is present the first configured Ext is used instead." , what happens if the call is on Ext 2, will that function works as well? If I understands, the call parking will only work for calls located at Ext 1 only if used vid=1. M i correct?

As for the hold and problem with Asterisk, here is what I m confused. Call parking in Asterisk requires me to type *2 then 70, or *270. *2 is the parking fonction, while 70 is the park number. However, transferring a call is the same thing: *2 then extension you want to transfer. So I am not sure fi I am wrong, but I assume call transfer and call parking is the same, only the extension varies.. Could you give me a hunt where to look into hold fonction in asterisk/incrediblepbx? Is it related into Park?

[Dan Lukes]

you've been very helpful!

Then consider rating valuable responses. ;-)

what happens if the call is on Ext 2, will that function works as well? 

SpeedDial has no relationship to current calls. SO same things will happen regardless there is a call on Ext 2 or there is no call at all. SpeedDial will hold current call (if any) and will initiate brand new call to number 70 in the context of Ext 1 configuration (if vid=1) or in the context of first configured Ext (if there is no vid parameter.

Call parking in Asterisk requires me to type *2 then 70, or *270

Yes, but Asterisk require you use DTMF to place such order. There is no way to configure Line key to put DTMF digits into an active call.

*2 is the parking function, while 70 is the park number. However, transferring a call is the same thing: *2 then extension you want to transfer. 

We are off-topic here a lot - it's forum dedicated to SPA IP Phone, not the Asterisk. You should ask Asterisk-only related question on an Asterisk-related forum.

But in short ...

In Asterisk world a "park" request is ordered in the form of "forward to special number. So the "park request" and "forward request" are similar.

But real world is more complicated. Park feature is provided by Asterisk only and ot can be requested by DTMF only. Forward is provided by Asterisk as well, but it can be requested either by DTMF or by dedicated SIP request sent by phone. Both attended and blind transfer can be requested by phone. There are [xfer] and [bXfer] PSK shown on phone's display (if there's active call only) dedicated for such kind of request.

Could you give me a hint where to look into hold function in asterisk/incrediblepbx? Is it related into Park?

It depend on overall concept of PBX configuration. No generic advice possible here. Increase verbosity and debug level of console - I'm sure there will be a notice related to issue. The context of such message (e.g. lines preceding and following it ) will give us the hint.

[ncixbrian]

Done for rating, just noticed I can rate answers :)

I have some trouble to fully understand everything, only because I have trouble to understand some words.

What does "in context of ext1" means?

Not sure what is DRMF, something like Dual-tone multi-frequency signaling, that is what I found on google.

Do you know if I need to put some configuration speed dial in Asterisk as well, or only cisco IP phone is enough? If I understand, speed dial is like "shortcut codes or functions" such as a shortcut to call a phone number ?

[Dan Lukes]

It may be caused by either my poor English or because you are not familiar with some basics. Or by both. Don't give up and try ... ;-)

What does "in context of ext1" means?

There are eight SIP configuration on SPA508G - eight Ext N tabs in WWW UI. E.g. up to eight different upstream switches the phone can register and/or place calls to. An outgoing call, like the one triggered by SpeedDial, need to decide which configuration (e.g. upstream switch) needs to be used for the purpose of such call. So saying "call is placed in context of Ext 1" I mean the configuration of Ext 1 is used for the purpose of such particular call.

Not sure what is DRMF

Yes, it is Dual Tone Multi-Frequency signaling. It is term coming from analog telephony era. Despite the SPA504G is based on SIP, understanding of analog phony principles will help you to understand some aspects of SIP telephony.

Do you know if I need to put some configuration speed dial in Asterisk as well, or only cisco IP phone is enough? If I understand, speed dial is like "shortcut codes or functions" such as a shortcut to call a phone number ?

You hit!. Speed Dial IS shortcut to call a phone number. Nothing more.

If you have implemented a call processing in Asterisk already you need change nothing. Asterisk is not aware about exact method you used to initiate new call on phone. It's just new incoming call that needs to be processed. Asterisk doesn't care you placed digits one by one by your finger or you used speed-dial button to place it at one.

But it seems you are still thinking about the call parking initiated via Speed-Dial button. So don't be confused. Speed Dial button configured on SPA508G can initiate brand new call only. Yes, the destination number may be special number used as "supplemental feature code", but the phone is not aware of it. It's Asterisk sovereignty to recognize feature code and process the call accordingly.

You claimed the Asterisk can recognize *2 DTMF signal to initiate transfer. Yes, but it doesn't mean you can call *2 as destination number. Such attempt will be processed like casual call to *2 number.  If you wish the *2 will be recognized as feature code, you need to implement it.

Asterisk expert in appropriate forum will give you better answer, but I assume you will not be able to:

1. identify other call from same phone (remember - SD will put current call on hold and place brand new call to the number configured)

2. even if you will identify the call in question, I know no way how to affect other unrelated call on Asterisk

Note "I know no way" doesn't mean "it's not possible". Better to ask on an Asterisk related forum.

[ncixbrian]

Hey Dan,

I have a quick 2 questions,

1) Asterisk does not support shared line (system key), but using cisco IP phones, is it possible to connect those phones to asterisk but making the phones act for a key system? Or "key system"/SLA is only configurable or supported by PBX and has nothing to do with IP phones? Will other IP phones other than cisco able to do the job?

2) Again about key system, what will happen if you put the same extension (for example 101) for both phones? Can key system be used?

[Dan Lukes]

I'm not sure what you mean saying "system key". So I don't know what you are asking for.

Also, I'm not sure what feature you expect from shared line. There is no problem to create shared lines on Asterisk. We use it.

According [2] - I still don;t know what's the "key system". But thre is no problem to assign extension to any number of physical device.

Just simple fragment of configuration:

exten => 100,n,Dial(SIP/DEV-FirstSPA504G&SIP/DEV-SecondSPA504G)

a call to extension 100 will be dispatched to both devices at the same time. They will ring simultaneously and if a line will pickup up the call, second line stop ringing. It is so called shared line.

[ncixbrian]

oh I meant key system.

For example, what I need is:

A call comes in, and it rings on all the phones, then phone a put on hold, and tell phone B to take the call. B resume the line from phone A. 

I know asterisk doesn't offer that feature, or it only could be done via SLA "trunk as line", however SLA has numerous limitations.

The way to bypass this problem looks to be more complicated, such as making it work in an indirect way. Here is what it could be done (some way to bypass, i used my imagination):

1- BLF + Call parking + Call pick up. Limitation = need more key lines to see parking slot, call dropped if parking slot if full, call dropped if accidentally pressing on the parking slot (I posted it in an Asterisk forum hoping to find an answer...)

2- Creating a queue for each phone extension. Phone A has queue A, phone B has queue B, etc.. therefore if Phone A wants Phone B to take the line and if ever Phone A is busy, phone A transfert the call to Queue B. Limitation = I don't know if it's possible for Phone A to see if any calls are waiting in queue (as the call will not be monitored on Phone B anymore, it is therefore monitored by no phone while placed in queue), impossible for Phone B to choose a specific call from the queue if there are many slots placed in that queue. 

3- Phone A transfert the call to Phone B extension, while setting timeout ring to Queue B.

So I was wondering if sharing an extension on different phones will allow any phone to resume the call, without using any of the 3 methods above. As far as i know, there is no way for Asterisk to make it happen. It is confirmed with my testings of Softphones. However, now that I have IP phone (not softphones), I wondered if IP phones are able to resume a call from another IP phone without call parking by doing some magic configuration on the phones instead without any change of Asterisk.

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After posting this in asterisk forum http://forums.asterisk.org/viewtopic.php?f=1&t=92571

They told me the problem with call failure is rather a phone problem and not asterisk :S  I tried on softphone, and somehow, there is no call failure when parking slot is full for example, but on IP phones, it's a different story.