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Any way to get Message Waiting Notification from the PSTN line?

Started by dani, April 13, 2012, 09:37:53 PM

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dani

I just realized that if someone leaves a message at our PSTN (LINE port) the visual and audible notifications are not forwarded to the PHONE port.

Is there ANY way to know if there is a message waiting, besides pressing #?

Thanks

RonR


dani

Thanks for the info.

That's the first big "BUMMER" i bump into, with this unit.

Can it detect the "stutter" tone and notify it somewhere in its web interface?

I tried monitoring the LINE port status, and left a voicemail but the loopcurrent,vbat and tipringvoltage settings didn't move.

If the stutter got indicated somewhere in the web interface, it could be scrapped.

RonR


dani

Thanks.

"Selling" this toy at home just got a lot harder.

Dani

infin8loop

I put a voice mailbox at voip.ms (on SP2) on my PSTN line by setting:

Physical Interfaces -> Line Port -> Calling Features ->
CallForwardOnNoAnswerEnabled : checked
CallForwardOnNoAnswerNumber : sp2(0118835100xxxxxxxx)
CallForwardOnNoAnswerRingCount : 5

Where "0118835100xxxxxxxx"  is a free iNum DID provided by voip.ms.  In the menu at voip.ms  DID Numbers -> CallerID Filtering I have a rule for inbound from 0118835100xxxxxxxx to immediately go to the voice mailbox (also on failover conditions Busy, No Answer, and Unreachable).  The originating callerid number is lost in the forward to voip.ms (which actually makes this Rube Goldberg work) so I have "say callerid" turned off in the voip.ms mailbox configuration.  MWIEnable and X_VMWIEnable are checked on SP2. I turned the answering machine off on the base unit of the cordless system connected to the phone port. I have speed dial #2 set to **2*97 to retrieve voice mail from voip.ms. There is no charge from voip.ms for this forwarding because it's voip.ms DID to DID dialing (the iNum is effectively calling itself.. think about that for a second and it may still not make sense to you, a drink might help).   I get the stutter VM waiting sound but I guess my cordless doesn't support the visual indicator. I probably couldn't see the indicator anyway. Since I configure my OBi locally (No, I'm not brainwashed nor a cult member although perhaps newbeeish at times and slowbeeish at others) I have speed dial #1 set to my google voice number because dialing 1# and 2# seems intuitive to me to retrieve VM from SP1 and SP2 respectively. The ObiTalk portal would force speed dial #1 to a soft phone that I don't ever use (unless I'm experimenting).

This method of madness probably (maybe?) could be adapted for use with voip providers other than voip.ms.
It's not exactly what you wanted or needed, but with the OBi, what is?!?

"This has not only been fun, it's been a major expense." - Gallagher

dani

infin8loop: yes, most of it makes sense and seems a good way to restore the natural order of things (at least to the way the phone users here are used to).

The only thing I'm not sure about is why caller id gets lost. Is it because it's an iNum? what if you forward to a sub acct or other kind of virtual sip?
In any case caller id is stored in the Obi history (which I poll and save on an hourly basis)

This may be a good excuse to shave the mailbox from the Bell phone bill. It could even pay for an extra real DID, if that would help in any way.

Thanks for the suggestion.

hwittenb

Quote from: dani on April 13, 2012, 11:39:32 PM
The only thing I'm not sure about is why caller id gets lost. Is it because it's an iNum? what if you forward to a sub acct or other kind of virtual sip?
Dani,
You can retain the caller id if you do the forwarding to voip.ms as a sip uri and you have set X_SpoofCallerID yes in the ITSP Profile for the SPx sip account, for example forward to SP2(5141231234@losangeles.voip.ms) or SP2(5141231234@sip.voip.ms).

Be aware that setting X_SpoofCallerID can cause other problems if you are using the automated attendent (AA) to bridge other calls and your voip provider does not allow a spoofed caller id. 

Edit: The reason the caller id gets "lost" is because the call is forwarded by voip.  Voip providers require authentication to terminate calls.  As part of authentication many/most voip providers who authenticate by userid and password do not allow "spoofed" caller id's.  They want to see the caller id that is represented by your account.  A few who authenticate by ip address are not that rigid.  You can setup a sub account under voip.ms that has an option to authenticate by a static ip address.

When you send a call by sip uri to a voip account it is like a normal incoming call that is not authenticated.  I believe that is why you can "spoof" the caller id on the call.

dani

Looks like I have some experiments to do. Thanks all for the suggestions.

infin8loop

For the method I suggested, if the original callerid number is retained then it won't match the iNum filter at voip.ms that sends the call immediately to voice mail. I believe this would create another inbound call back to the OBi and the caller would have to wait longer for that bridged part of the call not to be answered and then be put to voice mail.  I have not tested this but this looks like it could be adjusted by fiddling with the voip.ms DID setting "Dial Time Out in seconds" for the iNum which looks to be set at a ridiculous 60 seconds (12 rings. I don't recall setting this but maybe I did).  In our case I don't want to set it too low because the iNum is also used for inbound calls by some of my daughter's International friends from summer camp and it needs to be high enough for calls to be answered if there is someone here to answer them. I appreciate the feedback and dialog this has created!
"This has not only been fun, it's been a major expense." - Gallagher

hwittenb

infin8loop,

Looking at the voip.ms offerings, if you wanted to retain the incoming caller id on the forwarded call to voip.ms, I believe you could get a "Virtual SIP Number" from voip.ms and setup routing for all calls to the virtual sip number "DID" to go directly to your voip.ms voicemail.  On the OBi you would do your forwarding to the sip uri for your virtual sip number and also on the OBi setup X_SpoofCallerID for your SPx that you will use for the forwarding.  I haven't tested this, but I believe it would work.

I think the forwarding to the "virtual sip number" would also be fairly fast.

infin8loop

hwittenb,

The voip.ms "Virtual SIP Number" works great and the callerid is retained on calls sent to the mailbox when using the sip forward.  The virtual sip number is just 25 cents a month (less than a penny a day).  In this scenario it looks like there is no per minute charge levied. Otherwise It's only 1/10 cent per minute for virtual sip numbers.  I noticed the "virtual sip numbers" before but never realized why I would use one. It's always helpful when someone "connects the dots".

Great info.  Thanks.  

dani,

Hopefully this gives you another option to experiment with.  A voip.ms virtual DID URL in the form
sp2(11aaaaaaxxx@servercity.voip.ms) can be substituted for the iNum in my original post where "All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits.".
Using this method, no CallerID filter is needed at voip.ms. Just configure the virtual SIP DID to go to voice mail immediately.
"This has not only been fun, it's been a major expense." - Gallagher

dani

Been working on this:
Got the virtual sip connected to the mailbox, got the Obi to forward on No Answer. That part works fine.
Also programmed the "VM" button on the phone to dial **2*97. Success so far.

Not so successful:

- Even when ITSP Profile B -> SIP -> X_SpoofCallerID has an X, I still get my own Caller ID. Not sure why...

- just realized that replacing Bell Answering Service with a voip.ms voicemail works fine on NO ANSWER, but not if the PSTN port is BUSY. The ForwardOnBusy only works when the PHONE is busy but the LINE is free.
Far-from-ideal workarounds I can think of:
1) use voip.ms mailbox on busy and bell mailbox on busy (very awkward to manage, plus still paying BELL)
2) do not use the PSTN for outgoing local calls (expensive if going out through voip.ms, caller-id messes up if going out through GV)

Thanks all for your help.

hwittenb

Dani,

You don't have any way to know someone is calling your pstn line when it is busy, only the phone company knows that.  Of course you can hire them to forward somewhere on busy but the last time I checked my phone company wanted $7.50/month for that little feature.  Another choice is to add call waiting to your pstn line but that also costs money and gets a little complicated pressing the flash to the phone line to answer the call.

You can put your outgoing calls thru a different provider that costs less than voip.ms.  Localphone goes for 0.5c per minute. You can set your caller id with them.  I don't think their voice quality is quite as good as voip.ms.  You can configure both voip.ms and localphone on your adapter using one of the voice gateways for localphone.

As for the caller id on a sip uri call, I don't know what the problem is there.  I setup my OBi110 on the Line Port to forward on no answer to a sip uri and I monitored the outgoing voip call packets with WireShark packet trace.  The outgoing Sip Invite showed the CallerID in the sip INVITE header field in both the From: and the Remote Party ID: fields.  So in my case it is sending it. 

Of course the OBi has to receive the incoming caller id from the pstn line.  The OBi has a Call History function.  When you look at the Call History you should see the incoming call on the Line and the outgoing call for the Call Forward.  I initially thought that was a reliable way to tell if the original caller id went out with the forwarded call but it is not reliable.  The "Peer Number" shows whether or not it went out with the forwarded call.  The "Peer Number" is the incoming caller id.  I ran some tests with X_SpoofCallerID on the ITSP Profile B for my SP2 both enabled and disabled and monitored the results with WireShark.  That is the setting that does make the difference.

The OBi also has a syslog function that is almost worthless for debugging.  It will show, however, if an incoming caller id is detected on an incoming pstn line call on the Line Port, but the Call History also does that.

To run a WireShark trace you need to run the packets thru an old hub, not a switch, with both your pc and you OBi attached so that your pc can see all the packets going to/from the internet.  The hub is hard to find if you don't have one.

dani

The plot thickens:
I've been testing this, and found the following:

- the phone NR is overwritten, but the original NAME (if available) is forwarded.
- voip.ms is charging me for the incoming forwarded call. It looks like the Obi transfers the analog call so voip.ms considers it Inbound DID instead of Inbound SIP

- I also tried to forward to sp2(11xxxxx9999@sip.voip.ms) instead of @toronto.voip.ms. The mailbox behaved differently (didn't play my custom msg) and the VMI didn't work.

Are any of these settings related? Do you guys have the defaults on them?:
X_UseRefer
X_ReferAOR
X_Use302ToCallForward
X_InsertRemotePartyID
X_MWISubscribe
X_MWISubscribeURI

Thanks

Dani



MichiganTelephone

Quote from: dani on April 15, 2012, 05:29:27 PMAre any of these settings related? Do you guys have the defaults on them?:
X_UseRefer
X_ReferAOR
X_Use302ToCallForward
X_InsertRemotePartyID
X_MWISubscribe
X_MWISubscribeURI

You should be able to find information on those settings (and the defaults) in the OBi Device Administration Guide.
Inactive, no longer posting or responding to messages.  Goodbye and good luck.  Some of my old Obihai-related blog posts have been moved to http://tech.iprock.com - note this in NOT my blog; I have simply given the owner permission to repost some of my old stuff.

dani

Yes, been reading the Admin guide.
Just curious what settings hwittenb and infin8loop had when testing this, in case they were the reason my setup behaves differently.

THanks

infin8loop

Dani,

I goofed. Upon further examination I discovered my initial testing of the voip.ms virtual sip number was not valid.  However, in re-testing I do get a valid callerid number in the voip.ms mailbox for calls routed to the voip.ms mailbox by way of Line port CallForwardOnNoAnswerNumber : sp2(11aaaaaaxxx@servercity.voip.ms).  There is in fact a charge of .001 (1/10) of a cent per minute for this to happen.  The laundry list of Obi parameters you inquired about all have the Default box checked.

I apologize for my confusion. I have two OBi110 units each connected to a different PSTN line and subaccount at voip.ms. Apparently I need to draw myself a diagram of the setup. I hate it whan that happens. 
"This has not only been fun, it's been a major expense." - Gallagher

dani

infin8loop

I got the bit about the charge. It's part of the virtual sip.

But... is this (working) setup you describe any different from what we've been testing until now?

I have:
sp2(111xxxxx999@toronto.voip.ms) on LINE CallForwardOnNoAnswerNumber,
LINE CallForwardOnNoAnswerEnable clicked
ITSP Profile B X_SpoofCallerID clicked

It all works, except the forwarding of the CID.

THanks

RonR

Quote from: dani on April 15, 2012, 09:01:12 PM
I have:
sp2(111xxxxx999@toronto.voip.ms) on LINE CallForwardOnNoAnswerNumber,
LINE CallForwardOnNoAnswerEnable clicked
ITSP Profile B X_SpoofCallerID clicked

It all works, except the forwarding of the CID.

FWIW, I did some testing and CallerID is definitely passed from SP1.  I don't have CallerID service on my PSTN Line, so I can't test it.  The settings I used were:

Voice -> Services -> SP1 Service -> CallForwardOnNoAnswerEnable : (checked)
Voice -> Services -> SP1 Service -> CallForwardOnNoAnswerNumber : SP2(1777xxxxxxx@in.callcentric.com)
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : (checked)

With X_SpoofCallerID enabled, Callcentric receives the CallerID from SP1.  With X_SpoofCallerID disabled, it does not.  The difference was also obvious in the SIP headers captured with WireShark.