VoIP.ms Wiki - User contributions [en]

Call Recordings

2019-06-19T22:09:45Z

Bills: /* Recording Outgoing Calls */ sp latter -> later

This feature allows you to record your calls, whether they are incoming calls by enabling "Record calls" in the settings of your DID number, or outgoing calls by enabling "Record calls" in the settings of your Sub account.
The customer portal also offers a page that allows you to manage your recorded calls. This powerful feature not only gives you the option to listen the calls from the portal, but you can also download the recordings and send them to an email address.
This feature is the perfect solution for companies that wish to maintain recordings of calls made and/or received for quality purposes, or for our clients who simply want to record calls for security reasons.

__TOC__

== Enabling Call Recording ==

Call Recording can be enabled per DID basis for incoming calls and per SIP/IAX basis for outgoing calls.

=== Recording Incoming Calls ===

You have the option to enable this feature per DID number basis. It can be enabled at the moment of ordering a new DID number with the setting '''Record calls''', or by enabling it latter on, from the customer portal, on "DID numbers" Tab> "Manage DIDs" option> Edit DID> "Record calls" setting.

[[File:RecordingDIDs.png|thumb|none|800px]]

Note: When '''Record calls''' is set to '''yes''' for a DID number, all incoming calls to that DID number will be recorded, and stored accordingly.

When you enable '''Record calls''' for a DID number, a small microphone icon will show up next to the number, under the '''Options''' column, on the "Manage DID(s)" page. This visual indicator will help you to identify which DID numbers have the Call recording option enabled.

[[File:EnablediconDID.png|thumb|none|800px]]

=== Recording Outgoing Calls ===

Similar to the option for recording incoming calls, recording outgoing calls can be enabled per SIP/IAX account/sub account basis.

For the main account, head Main Menu> Account Settings> General, change the setting '''Record Calls''' to yes, then click on '''Apply'''.

[[File:RecordingMain.png|thumb|none|800px]]

For any sub account, it can be enabled at the moment you create it, or later on from the customer portal, on Sub Accounts> Manage Sub Accounts> Edit Sub Account, the setting is '''Record Calls''', set to yes to enable, scroll down to the bottom of the page and apply the change.

[[File:RecordingSubs.png|thumb|none|800px]]

Note: When '''Record calls''' is set to '''yes''' for a SIP/IAX account, all outgoing calls from that account will be recorded, and stored accordingly.

When you enable '''Record calls''' for a sub account, a small microphone icon will show up next to the sub account ID, under the '''Username''' column, on the "Manage Sub Accounts" page. This visual indicator will help you to identify which sub accounts have the Call recording option enabled.

[[File:RecordingIconSubs.png|thumb|none|800px]]

Additionally, you can find a visual indicator (microphone icon) of which SIP accounts have "Call Recording" enabled, including the main account, on the '''Portal Home''' (Main Menu> Portal home).

[[File:IconPortalHome.png|thumb|none|800px]]

== Managing the Recorded Calls ==

From the customer portal, you have access to any recorded call by our system for your accounts or numbers. To review your recordings click on the "DID Numbers">>"[https://www.voip.ms/m/call_recordings.php Call Recordings]" menu option. The page also offers a range of different options for you to search through your call recordings.

[[File:CallRecordingsPage.png|thumb|none|800px]]

'''Search Range'''- Is the time frame the system will search for stored recordings.

'''Account'''- Allows you to select the SIP/IAX account you want to search, to display only the recordings associated to the SIP/IAX in question. This option will only show the sub accounts that have had "Record Calls" option enabled and recorded at least one call.

Note: All the incoming calls are are logged as associated to the Main Account. Selecting an specific Sub account in this option, will make the portal to display only OUTGOING calls associated to the Sub Account selected.

'''Call Type'''- This option allows you to filter the calls by incoming calls, outgoing calls or both simultaneously.

In addition, there is a '''Search''' bar to find more specific results. The search option will match results in the "Date/Time", "Caller", "Destination" and "Account" columns.

Using the different filtering criteria, you are able to easily find specific recordings, knowing some information of the call, like the date, time, caller's number or destination.

Once you have results displaying in the page, you can listen the recording by clicking on "Play Audio".

In the '''Actions''' column, you have 3 different icons, each representing one option:

[[File:Actions column.png|thumb|none|800px]]

1.- '''Download'''. With this option you can download the recording of the call directly into your computer or device.

2.- '''Send email'''. This option allows you to send the recording as an attached file via email. Click on this icon and then type in the email address you want to send the recording to.

3.- '''Delete Recording'''. As the name refers, this option is to delete the recording. You will need to confirm the action by clinking on "Delete Recording" in the prompted window.

[[File:DeleteRecording.png|thumb|none|800px]]

Recordings downloaded or sent via email, will be in the format MP3, which offers great compatibility and is very small.

== Billing Call Recordings ==

All charges related to recorded calls, incoming or outgoing, will be included in the "Communication Charges" when creating an Invoice from the customer portal (Finances> Generate invoice). Please notice, "Communication charges" also includes the costs of the calls.

'''Important Note:'''
'''Recording calls has a cost of $0.0025 (0.25 cents) per minute.'''

You can also find more details of all your recordings, cost per recording, and total cost of all the recordings in a period of time, in the Customer Portal's CDR, on CDR and Reports> Call Detail Records.

To show only the records for Call recordings, type the words "Call Recording" in the search bar of the CDR.

Charges generated by recorded incoming calls will show up with the description "Inbound Call Recording". In order to show the recordings of incoming calls on the CDR, the '''Call Type''' must be set to '''All Calls''' or '''Incoming Calls: All'''.

Charges generated by recorded outgoing calls will show up with the description "Outbound Call Recording". In order to show the recordings of Outgoing calls on the CDR, the '''Call Type''' must be set to '''All Calls''' or '''Outgoing Calls: All'''.

[[File:RecordingsCDRwIndictations.png|thumb|none|800px]]

*1.- Total cost per individual recording, based on the per minute rate ($0.0025) and the time.

*2.- Total cost for all the recordings during the period of time selected.

== Legal Disclaimer ==

'''VoIP.ms''' offers you the possibility to record your phone conversations, to your entire and sole discretion. You should be aware that the laws regarding the notice and notification requirements, legality and use of such recorded conversations vary from country to country, state to state or province to province. As such, your recording of such conversation might be illegal, therefore you should make the proper verifications. 
By requesting or using the recording feature, you acknowledge and agree to be solely and entirely responsible for complying with all international, federal, state, provincial and local laws and regulations in the relevant jurisdiction when using this feature, and you warrant to '''VoIP.ms''' that you are acting legally. You expressly acknowledge and agree that '''VoIP.ms''' will not and cannot be held liable for any and all claim, directly and indirectly related to your recording of phone conversations. Without limitation to the Terms of Service, you agree to release, indemnify and to hold harmless '''VoIP.ms''' from and against any and all claims, damages or liabilities of any kind related directly or indirectly to the recording of any phone conversation using the Service.

Talk:Grandstream HandyTone 702 - HT702

2016-12-29T04:46:39Z

Bills: Add to earlier note - preferred dtmf in-audio very important.

These instructions appear to apply to the 'Grandstream HandyTone HT802', just fine, too.

(Websites: HT702 Product Page, http://www.grandstream.com/index.php/products/ip-voice-telephony/consumer-analog-telephone-adaptors/ht702_704 - broken. Try http://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/handytone-702/704 ?)

Websites: HT802 Product Page http://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802

Help / Support: Grandstream Support http://www.grandstream.com/support/resources/?title=HT802

Re: "Please go to your Graphical user interface and go to 'Advanced Settings' tab and look for "Firmware Upgrade and Provisioning" and disable it. " - more details as to how to do that would be useful. There are many on the page. e.g. If setting 'Automatic Upgrade:' to no (which appears to be the default) does it, say so here.

I (purchased 12/9/16) appear to have received firmware 1.0.1.9, current appears to be 1.0.2.5 Firmware server path comes as 'fm.grandstream.com/gs' and per web page should be 'firmware.grandstream.com'. I ended up upgrading firmware via local download, so don't know that merely changing the web page works - e.g. Change web page, upgrade once, set automatic upgrades off.

The current graphic is: (a) too small, e.g. can't read dial plan to apply it (perhaps a text dump would be useful). Default dial plan is: '{ x+ | \+x+ | *x+ | *xx*x+ }' - useful to note on this page so the reader can go back to defaults if something not working; (b) quite a bit different from the 802 one.

If you send me a destination, I can send 802's config pages.

Note the current graphic shows DNS mode SRV, but text afterwards says use A - should update graphic. (I followed the graphic, got to text later.)

"If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid green color, then your unit is configured and ready to make calls." - the 802 does not have the 'traditional' lights. Here, there are individual lit symbols for each phone. (See the web page.)

I've only just gotten / configured the 802, but initial tests to/from vm appear fine. For the wiki - it's a start.

--[[User:Bills|BillS]] 04:18, 10 December 2016 (EST)

Further to the above ...

I found when calling the CRA that tones were not getting sent to their IVR, despite having applied the 'Preferred DTMF method:' settings per the wiki page. Reviewing the settings now, I note immediately below those settings, 'Disable DTMF Negotiation:' - set to No as per the wiki page. Setting it to yes ['Yes (use above DTMF order without negotiation)'] appears to have resolved the issue.

[[User:Bills|BillS]] 11:21, 21 December 2016 (EST)

Then I found I could not get into vm to retrieve messages - until I set, as per page, Preferred DTMF method: In-audio, for all 3 settings. (A purchaser of this unit will be most unhappy, I think, out of the box - so a separate wiki page for this device will be very, very, useful.)
[[User:Bills|BillS]] 23:46, 28 December 2016 (EST)

Talk:Grandstream HandyTone 702 - HT702

2016-12-21T16:21:41Z

Bills: Add to note re: 802 to disable DTMF negotiation.

Talk:Grandstream HandyTone 702 - HT702

2016-12-10T09:18:41Z

Bills: Thes 702 instructions appear good for 802, too.

Talk:SIP URI

2016-10-18T13:17:27Z

Bills: /* On Rev.: "We have been notified that SIP Broker service is not working correctly." */

Some things could use some clarification.

STRESS that unless one has a DID, SIP dialing TO voip.ms does not work. (This caught me, and I've been around for awhile!)

Using JohnSmith@ is perhaps a misnomer for this article / voip.ms - try 123456@ instead.

Perhaps give examples / show difference of calling an account, and a subaccount, where subaccount mentioned. e.g. 123456@ vs 1234567@ vs 123456_subaccount name.

Stress / split out use of SIP URI settings as phone book settings vs other uses, and IN PARTICULAR, that the voip.ms sip uri is entered on the OTHER system.

"Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms." - this isn't quite true, so split it out. e.g. Phone book. Note that SIP URIs are noted on portal page and in 'i' of subaccount management.

"You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s). " add ... on the other company's system. Again, youraccount@ is a misnomer for voip.ms use, to which this wiki pertains. Use 123456@voip.ms instead.
The quote also mixes from's and to's. Perhaps rephrase to: From other companies you can dial your DID numbers using a SIP URI address, such as 123456@voip.ms. You will need to create such a string at the other company.

" Creating a new SIP URI

To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section". " is confusing.

Split out phone book use from non. Move phone book use up to this point in article. i.e. calling out. Then separately deal with calling in separately.

"use the {DID} tag" is a bit mystifying. Perhaps use an example first, e.g. 8005551212@voip.ms. Then move on to show how using {DID}@voip.ms would automatically substitute in the 8005551212 for you.

" 1{DID}@128.144.122.12
12143221234@128.144.122.12
some_extension_name@128.144.122.12:5080
other_extension_name@voip.example.com
extension_name@123456_subaccount
{DID}@123456_subaccount (This produces the same result as routing the call directly to the sub account.) "
is confusing, probably because of the mix of incoming and outgoing in the same section.

The implication is that one can make up a random SIP URI, put it in the phone book, and external callers will automatically be routed. Yet that seems unlikely / is not confirmed / by example.

"{DID}@123456_subaccount" is particularly perplexing, as other references would say that it should be 123456_subaccount@toronto3.voip.ms - the implication here is that 18005551212@123456_subaccount@voip.ms is a viable address.

"Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call)." - this does not address incoming call. Moreover, as far as I know, default SIP URIs are auto-generated. And users cannot create their own. Else, what happens with two voip.ms John Smith users when they each try to create JohnSmith@voip.ms.

Thanks for listening.

-- Bill
[[User:Bills|BillS]] 12:08, 17 February 2015 (EST)

== On Rev.: "We have been notified that SIP Broker service is not working correctly." ==

Received the notice of page change, upon addition of "We have been notified that SIP Broker service is not working correctly.", etc.

In checking it out I observed the "This format of SIP address is used by services such as wikipedia:SIP Broker as a means to reach voip.ms subscribers." text.

Clicking on the wikipedia link, I got "This page has been deleted."

The (wikipedia) link should probably be substituted with something else.

-- [[User:Bills|BillS]] 09:15, 18 October 2016 (EDT)

Talk:SIP URI

2016-10-18T13:16:39Z

Bills: /* On Rev.: "We have been notified that SIP Broker service is not working correctly." */

Some things could use some clarification.

STRESS that unless one has a DID, SIP dialing TO voip.ms does not work. (This caught me, and I've been around for awhile!)

Using JohnSmith@ is perhaps a misnomer for this article / voip.ms - try 123456@ instead.

Perhaps give examples / show difference of calling an account, and a subaccount, where subaccount mentioned. e.g. 123456@ vs 1234567@ vs 123456_subaccount name.

Stress / split out use of SIP URI settings as phone book settings vs other uses, and IN PARTICULAR, that the voip.ms sip uri is entered on the OTHER system.

"Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms." - this isn't quite true, so split it out. e.g. Phone book. Note that SIP URIs are noted on portal page and in 'i' of subaccount management.

"You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s). " add ... on the other company's system. Again, youraccount@ is a misnomer for voip.ms use, to which this wiki pertains. Use 123456@voip.ms instead.
The quote also mixes from's and to's. Perhaps rephrase to: From other companies you can dial your DID numbers using a SIP URI address, such as 123456@voip.ms. You will need to create such a string at the other company.

" Creating a new SIP URI

To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section". " is confusing.

Split out phone book use from non. Move phone book use up to this point in article. i.e. calling out. Then separately deal with calling in separately.

"use the {DID} tag" is a bit mystifying. Perhaps use an example first, e.g. 8005551212@voip.ms. Then move on to show how using {DID}@voip.ms would automatically substitute in the 8005551212 for you.

" 1{DID}@128.144.122.12
12143221234@128.144.122.12
some_extension_name@128.144.122.12:5080
other_extension_name@voip.example.com
extension_name@123456_subaccount
{DID}@123456_subaccount (This produces the same result as routing the call directly to the sub account.) "
is confusing, probably because of the mix of incoming and outgoing in the same section.

The implication is that one can make up a random SIP URI, put it in the phone book, and external callers will automatically be routed. Yet that seems unlikely / is not confirmed / by example.

"{DID}@123456_subaccount" is particularly perplexing, as other references would say that it should be 123456_subaccount@toronto3.voip.ms - the implication here is that 18005551212@123456_subaccount@voip.ms is a viable address.

"Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call)." - this does not address incoming call. Moreover, as far as I know, default SIP URIs are auto-generated. And users cannot create their own. Else, what happens with two voip.ms John Smith users when they each try to create JohnSmith@voip.ms.

Thanks for listening.

-- Bill
[[User:Bills|BillS]] 12:08, 17 February 2015 (EST)

== On Rev.: "We have been notified that SIP Broker service is not working correctly." ==

Received the notice of page change, upon addition of "We have been notified that SIP Broker service is not working correctly.", etc.

In checking it out I observed the "This format of SIP address is used by services such as wikipedia:SIP Broker as a means to reach voip.ms subscribers." text.

Clicking on the wikipedia link, I got "This page has been deleted."

The (wikipedia) link should probably be substituted with else.

-- [[User:Bills|BillS]] 09:15, 18 October 2016 (EDT)

Talk:SIP URI

2016-10-18T13:16:18Z

Bills: /* On Rev.: "We have been notified that SIP Broker service is not working correctly." */

Some things could use some clarification.

STRESS that unless one has a DID, SIP dialing TO voip.ms does not work. (This caught me, and I've been around for awhile!)

Using JohnSmith@ is perhaps a misnomer for this article / voip.ms - try 123456@ instead.

Perhaps give examples / show difference of calling an account, and a subaccount, where subaccount mentioned. e.g. 123456@ vs 1234567@ vs 123456_subaccount name.

Stress / split out use of SIP URI settings as phone book settings vs other uses, and IN PARTICULAR, that the voip.ms sip uri is entered on the OTHER system.

"Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms." - this isn't quite true, so split it out. e.g. Phone book. Note that SIP URIs are noted on portal page and in 'i' of subaccount management.

"You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s). " add ... on the other company's system. Again, youraccount@ is a misnomer for voip.ms use, to which this wiki pertains. Use 123456@voip.ms instead.
The quote also mixes from's and to's. Perhaps rephrase to: From other companies you can dial your DID numbers using a SIP URI address, such as 123456@voip.ms. You will need to create such a string at the other company.

" Creating a new SIP URI

To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section". " is confusing.

Split out phone book use from non. Move phone book use up to this point in article. i.e. calling out. Then separately deal with calling in separately.

"use the {DID} tag" is a bit mystifying. Perhaps use an example first, e.g. 8005551212@voip.ms. Then move on to show how using {DID}@voip.ms would automatically substitute in the 8005551212 for you.

" 1{DID}@128.144.122.12
12143221234@128.144.122.12
some_extension_name@128.144.122.12:5080
other_extension_name@voip.example.com
extension_name@123456_subaccount
{DID}@123456_subaccount (This produces the same result as routing the call directly to the sub account.) "
is confusing, probably because of the mix of incoming and outgoing in the same section.

The implication is that one can make up a random SIP URI, put it in the phone book, and external callers will automatically be routed. Yet that seems unlikely / is not confirmed / by example.

"{DID}@123456_subaccount" is particularly perplexing, as other references would say that it should be 123456_subaccount@toronto3.voip.ms - the implication here is that 18005551212@123456_subaccount@voip.ms is a viable address.

"Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call)." - this does not address incoming call. Moreover, as far as I know, default SIP URIs are auto-generated. And users cannot create their own. Else, what happens with two voip.ms John Smith users when they each try to create JohnSmith@voip.ms.

Thanks for listening.

-- Bill
[[User:Bills|BillS]] 12:08, 17 February 2015 (EST)

== On Rev.: "We have been notified that SIP Broker service is not working correctly." ==

Received the notice of page change, upon addition of "We have been notified that SIP Broker service is not working correctly.", etc.

In checking it out I observed the "This format of SIP address is used by services such as wikipedia:SIP Broker as a means to reach voip.ms subscribers." text.

Clicking on the wikipedia link, I got "This page has been deleted."

The (wikipedia) link should probably be substituted with else.

[[User:Bills|BillS]] 09:15, 18 October 2016 (EDT)

Talk:SIP URI

2016-10-18T13:15:53Z

Bills: /* On Rev.: "We have been notified that SIP Broker service is not working correctly." */ new section

Some things could use some clarification.

STRESS that unless one has a DID, SIP dialing TO voip.ms does not work. (This caught me, and I've been around for awhile!)

Using JohnSmith@ is perhaps a misnomer for this article / voip.ms - try 123456@ instead.

Perhaps give examples / show difference of calling an account, and a subaccount, where subaccount mentioned. e.g. 123456@ vs 1234567@ vs 123456_subaccount name.

Stress / split out use of SIP URI settings as phone book settings vs other uses, and IN PARTICULAR, that the voip.ms sip uri is entered on the OTHER system.

"Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms." - this isn't quite true, so split it out. e.g. Phone book. Note that SIP URIs are noted on portal page and in 'i' of subaccount management.

"You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s). " add ... on the other company's system. Again, youraccount@ is a misnomer for voip.ms use, to which this wiki pertains. Use 123456@voip.ms instead.
The quote also mixes from's and to's. Perhaps rephrase to: From other companies you can dial your DID numbers using a SIP URI address, such as 123456@voip.ms. You will need to create such a string at the other company.

" Creating a new SIP URI

To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section". " is confusing.

Split out phone book use from non. Move phone book use up to this point in article. i.e. calling out. Then separately deal with calling in separately.

"use the {DID} tag" is a bit mystifying. Perhaps use an example first, e.g. 8005551212@voip.ms. Then move on to show how using {DID}@voip.ms would automatically substitute in the 8005551212 for you.

" 1{DID}@128.144.122.12
12143221234@128.144.122.12
some_extension_name@128.144.122.12:5080
other_extension_name@voip.example.com
extension_name@123456_subaccount
{DID}@123456_subaccount (This produces the same result as routing the call directly to the sub account.) "
is confusing, probably because of the mix of incoming and outgoing in the same section.

The implication is that one can make up a random SIP URI, put it in the phone book, and external callers will automatically be routed. Yet that seems unlikely / is not confirmed / by example.

"{DID}@123456_subaccount" is particularly perplexing, as other references would say that it should be 123456_subaccount@toronto3.voip.ms - the implication here is that 18005551212@123456_subaccount@voip.ms is a viable address.

"Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call)." - this does not address incoming call. Moreover, as far as I know, default SIP URIs are auto-generated. And users cannot create their own. Else, what happens with two voip.ms John Smith users when they each try to create JohnSmith@voip.ms.

Thanks for listening.

-- Bill
[[User:Bills|BillS]] 12:08, 17 February 2015 (EST)

== On Rev.: "We have been notified that SIP Broker service is not working correctly." ==

Received the notice of page change, upon addition of "We have been notified that SIP Broker service is not working correctly.", etc.

In checking it out I observed the "This format of SIP address is used by services such as wikipedia:SIP Broker as a means to reach voip.ms subscribers." text.

Clicking on the wikipedia link, I got "This page has been deleted."

The (wikipedia) link should probably be substituted with else.
[[User:Bills|BillS]] 09:15, 18 October 2016 (EDT)

Talk:Call Detail Records

2015-02-28T17:11:06Z

Bills: Date won't sort ascending.

== Date won't sort ascending. ==

'Blue Arrows: The blue arrows on the CDR allow you to sort each field in ascending order.' - doesn't appear to be entirely true, at least for date. Clicking on the arrow still returns the data in descending order. [Would be nice if clicking it gave chronological order, though!]
[[User:Bills|BillS]] 12:11, 28 February 2015 (EST)

Talk:DISA

2015-02-19T12:43:50Z

Bills: /* Second call? */ new section

It would be useful if this page made reference to, or used examples of, incoming toll free and outgoing toll, both local and long distance. I would guess that this is the more typical use case, at least of DISA.

If there is no difference in cost between going out to a local or long distance destination, then so note - although, perhaps, only needed on the page containing the referenced link.

I expect this costing example would be applicable to several wiki pages, thus a link may be the most appropriate way of handling this.

== Second call? ==

It would be useful to note if, or how, second calls can be made, or whether one must hang up and disa in again.

e.g. Suppose I call in, *97 for voice mail, then want to continue with the line and make another call. What is necessary to do that?

Flash or blind transfer wouldn't seem to be useful / make sense here. What does?

Cisco/Linksys Star Codes

2015-02-17T19:29:40Z

Bills: Remove the confusing '*'s in front of the codes, added internal speed dial access method.

The Cisco/Linksys ATA devices come with In-Built star codes that activate certain features on your device. These features are '''independent''' from the features that you can setup from the customer portal, however these will usually be supported by our service.

The following list will provide the most commonly used asterisk codes from your ATA device:

'''*69''' '''Call Return Code''' ''This code calls the last caller.''

'''*07''' '''Call Redial Code''' ''Redials the last number called. (Not in pap2t)''

'''*98''' '''Blind Transfer Code''' ''Begins a blind transfer of the current call to the extension specified after the activation code.''

'''*66''' '''Call Back Act Code''' ''Starts a callback when the last outbound call is not busy.''

'''*86''' '''Call Back Deact Code''' ''Cancels a callback.''

'''*05''' '''Call Back Busy Act Code''' ''Starts a callback when the last outbound call is busy. (Not in pap2t)''

'''*72''' '''Cfwd All Act Code''' ''Forward s all calls to the extension specified after the activation code.''

'''*73''' '''Cfwd All Deact Code''' ''Cancels call forwarding of all calls.''

'''*90''' '''Cfwd Busy Act Code''' ''Forwards busy calls to the extension specified after the activation code.''

'''*91''' '''Cfwd Busy Deact Code''' ''Cancels call forwarding of busy calls.''

'''*92''' '''Cfwd No Ans Act Code''' ''Forwards no-answer calls to the extension specified after the activation code.''

'''*93''' '''Cfwd No Ans Deact Code''' ''Cancels call forwarding of no-answer calls.''

'''*63''' '''Cfwd Last Act Code''' ''Forwards the last inbound or outbound calls to the extension specified after the activation code.''

'''*83''' '''Cfwd Last Deact Code''' ''Cancels call forwarding of the last inbound or outbound calls.''

'''*60''' '''Block Last Act Code''' ''Blocks the last inbound call.''

'''*80''' '''Block Last Deact Code''' ''Cancels blocking of the last inbound call.''

'''*64''' '''Accept Last Act Code''' ''Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.''

'''*84''' '''Accept Last Deact Code''' ''Cancels the code to accept the last outbound call.''

'''*56''' '''CW Act Code''' ''Enables call waiting on all calls.''

'''*57''' '''CW Deact Code''' ''Disables call waiting on all calls.''

'''*71''' '''CW Per Call Act Code''' ''Enables call waiting for the next call.''

'''*70''' '''CW Per Call Deact Code''' ''Disables call waiting for the next call.''

'''*67''' '''Block CID Act Code''' ''Blocks caller ID on all outbound calls.'' (''Note: This differs from the implementation of *67 by some landline providers in that the block remains active until turned off with *68'')

'''*68''' '''Block CID Deact Code''' ''Removes caller ID blocking on all outbound calls.''

'''*81''' '''Block CID Per Call Act Code''' ''Blocks caller ID on the next outbound call.''

'''*82''' '''Block CID Per Call Deact Code''' ''Removes caller ID blocking on the next inbound call.''

'''*77''' '''Block ANC Act Code''' ''Blocks all anonymous calls.''

'''*87''' '''Block ANC Deact Code''' ''Removes blocking of all anonymous calls.''

'''*78''' '''DND Act Code''' ''Enables the do not disturb feature.''

'''*79''' '''DND Deact Code''' ''Disables the do not disturb feature.''

'''*65''' '''CID Act Code''' ''Enables caller ID generation.''

'''*85''' '''CID Deact Code''' ''Disables caller ID generation.''

'''*25''' '''CWCID Act Code''' ''Enables call waiting, caller ID generation.''

'''*45''' '''CWCID Deact Code''' ''Disables call waiting, caller ID generation.''

'''*26''' '''Dist Ring Act Code''' ''Enables the distinctive ringing feature.''

'''*46''' '''Dist Ring Deact Code''' ''Disables the distinctive ringing feature. The default is *46.''

'''*74''' '''Speed Dial Act Code''' ''Assigns a speed dial number.''

'''*16''' '''Secure All Call Act Code''' ''Makes all outbound calls secure.''

'''*17''' '''Secure No Call Act Code''' ''Makes all outbound calls not secure.''

'''*18''' '''Secure One Call Act Code''' ''Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)''

'''*19''' '''Secure One Call Deact Code''' ''Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)''

'''''Conference Act Code''''' ''If this code is specified, the user must enter it before dialing the third party for a conference call. (Your conference call organizer provides you the code.)''

'''''Attn-Xfer Act Code''''' ''If the code is specified, the user must enter it before dialing the third party for a call transfer. (Your telephone administrator must provide you with this code, enabling you to transfer your caller to an outside number.)''

'''*99''' '''Modem Line Toggle Code''' ''Toggles the line to a modem. Modem pass-through mode can be triggered only by pre-dialing this code.''

'''*99''' '''FAX Line Toggle Code''' ''Toggles the line to a fax machine. (Not in pap2t)''

'''''Service Referral Codes''''' ''Codes used when you place your caller on hold and receive the second dial tone to blind transfer your caller. (Your telephone administrator must provide you with any such codes.)''

'''''Dial Feature Codes''''' ''Codes used when you first receive dial tone. (Your telephone administrator must provide you with any such codes.)''

'''''#''''' ''(2-9) Dial the internal (ata) specified speed dial number. (Add a '#' to end the interdigit timeout.)''

Help:Editing

2015-02-17T19:05:26Z

Bills: Created page with "See http://www.mediawiki.org/wiki/Help:Editing"

See http://www.mediawiki.org/wiki/Help:Editing

Talk:SIP URI

2015-02-17T17:08:46Z

Bills: Created page with "Some things could use some clarification. STRESS that unless one has a DID, SIP dialing TO voip.ms does not work. (This caught me, and I've been around for awhile!) Using JohnS..."

Talk:CallerID Filtering

2014-11-20T06:10:48Z

Bills: Comments to revision from 6030

Although filters and usage scenarios are well explained here, the steps to apply those filters is not evident here, unlike other articles.

So, for example, the usage scenarios are given, but not how to implement them. Screen shots of the did settings, and assigning a filter, would be useful.

In particular, the reason for my coming to this discuss page, how does one implement multiple filters? e.g. suppose one wants to have family cause all home extensions to ring at any hour, but if it is the in-laws calling send them directly to voicemail. No one pattern could cover all in-laws, so it would be useful if it were noted that only one pattern can be used, or how to effect multiple filters.
----
Added later: OK, I see my problem - unlike all other features, callerid filtering is not applied or indicated in any routing areas. In hindsight, this is obvious upon trying, and re-reading this article. However, for the new reader this is not particularly obvious.

So, could an example with multiple filters be added to this article? It would make the answer to the above question self-evident.
----
It would be useful to mention here if a hierarchy is applied, e.g. most specific to least. e.g. 800555* taking precedence over 800*, or even *, in terms of processing.
----
Re: Change from version 6030:
Thank you for the update. The use of the word 'priority' however, is confusing. That exact matches take priority is clear, however "then in priority from top to bottom" should probably be "then in order from top to bottom". The use of the word priority made me go look for how to set priority, to discover there is no ability to. Also, it would be useful if wildcard priority could be laid out, e.g. "519*" or "519xxxxxxx" - which takes priority, or even "519xxxxxx*". Thanks for the update - helpful.
----

Caller ID

2013-11-29T17:25:05Z

Bills: Update to correct some writing, phrasing, and language use.

Caller ID is a telephone service that transmits the calling party´s number to the called party´s telephone . When available the Caller ID number can be complemented with Caller ID name (e.g. John Smith)

If you are placing outgoing calls you will likely need to pass a Caller ID to ensure proper termination of your calls, particularly to properly reach toll free numbers.

There are two types of caller ID and it is important to understand that they are different things: Caller ID Number (CID), and Caller ID Name (CNAM).

Please note that the caller ID is only guaranteed while using premium routes, and only for US48 and Canadian calls. In Canada you may find caller ID working on some value routes.

Incoming Caller ID (from people calling you) is further described below.

__TOC__

== Outgoing Caller ID number ==

Caller ID number is the most common Caller ID type passed. If a more complex system capable of passing its own caller ID is being used, such as a [[Welcome#PBX|PBX]], the Caller ID field is likely set from the trunk, or one of its extensions.

If you are using devices like [[Welcome#Devices|Analogue Telephone Adapters]], [[Welcome#Devices|IP phones]], or [[Welcome#Softphones|softphones]], the caller ID number is available to be set from your voip.ms account via the customer portal.

To set the Caller ID number for your Main account, access "[[Account Settings]]" from the "Main Menu" menu, General Tab.

[[File:CIDmain.jpg|Main account Caller ID]]

The Caller ID number for a subaccount can be set at time of creation, or later by clicking Edit on the [[Sub Accounts|subaccount]].

'''It is strongly suggested a 10 digits Caller ID number be set to ensure proper call termination. The portal's Caller ID field only supports numerical digits.'''
'''Anonymous and Toll free caller IDs are not recommended, especially when Calling Toll free numbers, to reduce potential connection issues.'''

== Outgoing Caller ID name ==

The caller ID name is an additional information you can pass along with your Caller ID number. This will also be received on the callee's end and it could be your name or you Business.

For example: ''"John Smith"<9145551234>''

The sample above is a Caller ID, that includes both Caller ID name and Caller ID number, commonly abbreviated as CID and CNAM among other variations.

Is not possible to set any Caller ID name from the voip.ms portal.

If you will be making calls to Canadian numbers, you can simply pass the Caller ID name from your device or system (if it supports it, most [[Softphones]] do). You will need to check for a field on the interface from the device to enter this setting, and in case you are using a more advanced system, get assistance to set the outgoing caller ID name set up.

The Caller ID name on US however works different, this is controlled by a national CNAM database, with records of numbers and names matching each number.
When you make a call to a US number, you will send a caller ID number, and the system will check on the CNAM database for a name matching the same Caller ID number you passed, in order to display both name and number to final phone.

CNAM is only available for some USA numbers. In order to update your Caller ID on the CNAM database for your US calls, there is a process to follow which has a cost of $10 USD (one time only).
Please contact voip.ms support to get further details on what information you need to submit and to confirm if your local US DID is available for CNAM update.

'''CNAM update is only available for some Local US DIDs. Toll frees can not have their Caller ID name updated'''.

== Incoming Caller ID number and name ==

You will receive the Caller ID number and Caller ID name that the voip.ms server receives from the caller, this is exactly what will be sent to you on Incoming calls.
You can always check what Caller ID number voip.ms receives, by going into your [[Call Detail Records]] to check the incoming calls.

The incoming caller ID name, works almost the same way, except that this is an optional setting that you need to enable from per DID number on the [[Manage DID|DID settings]] page.
This option is called "Caller ID Lookup". When enabled, the system will perform a query on the LIBD/CNAM Database, for callers with Canadian or US CID number, in order to find a name matching that CID number.
The system then will display the result of this query in the Caller ID name portion of the '''Caller ID''', leading to a "Caller ID name"<5551231234> when people call your number.

'''IMPORTANT NOTE FOR CANADIAN DIDs'''

The majority of the Canadian DID numbers support CNAM Pass-Through. This means that for your incoming calls the system '''won't do a CNAM query''' (and not charge you either) if the incoming call is arriving already with a caller-id name, even if the DID receiving the call has the CNAM queries enabled in your VoIP.ms customer portal.
----

'''On a side note, outgoing Caller ID is not guaranteed on calls to Canadian cellular numbers, even when using the premium route. This is due to the way Canadian carriers work - they sometimes pass a random Caller ID that they have on record, changing the original.'''
'''This is out of our control as it is the way Canadian carriers handle calls to cellular numbers.'''

[[category:guides]]

Talk:CallerID Filtering

2013-09-06T11:20:53Z

Bills: Callerid Hierarchy?

Although filters and usage scenarios are well explained here, the steps to apply those filters is not evident here, unlike other articles.

So, for example, the usage scenarios are given, but not how to implement them. Screen shots of the did settings, and assigning a filter, would be useful.

In particular, the reason for my coming to this discuss page, how does one implement multiple filters? e.g. suppose one wants to have family cause all home extensions to ring at any hour, but if it is the in-laws calling send them directly to voicemail. No one pattern could cover all in-laws, so it would be useful if it were noted that only one pattern can be used, or how to effect multiple filters.
----
Added later: OK, I see my problem - unlike all other features, callerid filtering is not applied or indicated in any routing areas. In hindsight, this is obvious upon trying, and re-reading this article. However, for the new reader this is not particularly obvious.

So, could an example with multiple filters be added to this article? It would make the answer to the above question self-evident.
----
It would be useful to mention here if a hierarcy is applied, e.g. most specific to least. e.g. 800555* taking precedence over 800*, or even *, in terms of processing.
----

Talk:Time Conditions

2013-09-03T05:28:01Z

Bills: Reentrant?

- it's a little wiggy that time conditions can call time conditions. Explaining that, indeed explaining some of the reentrant paths (and under what conditions / why one might want to do so) would be useful - perhaps in a separate wiki article. 
- I assume one might like to call a time condition to drill down to more granular conditions. e.g. time condition 8 - 9 pm, to time condition 8:00 - 8:15, 8:15 - 8:30, and so on - assuming one ran out of time conditions. Discussing this in a paragraph would be useful.

Talk:Digital Receptionist (IVR)

2013-09-02T14:18:25Z

Bills:

- no indication in the text as to what the '*' digit does. 
- probably worth noting in text '#' is interdigit timeout. 
- I'm guessing an ivr with no digits, or only t and i digits, is a way to daisy chain recordings?

Talk:Digital Receptionist (IVR)

2013-09-02T14:16:14Z

Bills: Missing '*' digit explanation.

- no indication in the text as to what the '*' digit does.
- probably worth noting in text '#' is interdigit timeout.

Dial Plan for Linksys ATAs

2013-04-17T19:46:31Z

Bills: ... forgot an 's'.

The basic dial plan provided in the configuration samples for the Linksys ATA devices (like [[Cisco Linksys PAP2|PAP2]], [[Cisco Linksys PAP2T|PAP2T]] and [[Cisco SPA2100 Phone Adapter|SPA2100]]) should work with VoIP.ms without issue.

'''Voip.ms recommended dial plan''':

(911S0|310xxxx|<:1'''555'''>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

'''Note''': For 7 digit dialing, replace 555 by the area code of your choice.

This guide has been created to help you learn about Dial Plans. They can be customized according to your preferences. Please note that customizing your dial plan is entirely optional.
 

__TOC__

== What is a Dial Plan? ==

The dial plan is a string of characters that determines how entered phone digits are interpreted and transmitted by your ATA device. It also determines whether to accept, or reject, a call. A dial plan thus facilitates dialing, and also the blocking, of certain types of calls, such as long distance or international.

'''Please Note''': International calls can also be blocked within your VoIP.ms account.

== Digit Sequence ==

A dial plan contains a series of digit sequences, separated by the | character, entirely enclosed within parentheses. Each time a phone button is pressed, your ATA device will attempt to match the digit sequence in your dial plan.

{| class="wikitable"
|-
! Digit Sequence
! Function
|-
| style="text-align:center;"|0 1 2 3 4 5 6 7 8 9 * #
| You can use any of these characters to represent a pressed phone digit.
|-
| style="text-align:center;"| x
| Any phone digit.
|-
| style="text-align:center;"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *.
|-
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. For example, '''01.''' allows the user to enter 0, 01, 011 and so on.
|-
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The ''dialed'' digits can be zero or more characters. For example with this sequence '''<:1555>xxxxxxx''' if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press 6782345, the system transmits 15556782345.
|-
| style="text-align:center;"|, (comma)
| This can be used between digits to play an “outside line” dial tone after a user-entered sequence. For example, with this sequence '''9, 1x.''' an “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1
|-
| style="text-align:center;"|! (exclamation point)
| You can use this character to prohibit a dial sequence. For example with the sequence '''1900xxxxxxx!''' the system reject any sequence that starts with 1900.
|-
| style="text-align:center;"|S0 or L0
| Overrides the Short or Long inter-digit timer to 0 seconds.
|-
| style="text-align:center;"|P# (where # is the duration of the pause in seconds)
| Pauses # seconds.
|}

== Examples ==

Some examples of dial plan digit sequences:

* To dial any international number without using the 011 prefix.
<:011> [2-9]xxxxxxxx.

You can also accomplish this if you set the Dialing Mode to E164 in your [[Account Settings]]

* To block a call to a specific area code (replace 555 with the area code you want)
<:1> 555 xxxxxxx !

* The next sequence, allows you to dial your [[Phone book]] entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520
<:*75>xx< # : >

== See also ==
ATA's
* [[Cisco SPA2100 Phone Adapter]]
* [[Cisco SPA2102 Phone Adapter with Router]]
* [[Cisco Linksys PAP2]]
* [[Cisco Linksys PAP2T]]

IP Phones
* [[Cisco SPA504G Phone|Cisco SPA300/500-series 'phones]]
* [[Cisco Linksys SPA942 NA]]

[[Category:Guides]]

== References ==
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters]
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones]

[[Category: SIP devices]]

Dial Plan for Linksys ATAs

2013-04-17T19:43:17Z

Bills: ... and I run out of steam ...

The basic dial plan provided in the configuration samples for the Linksys ATA devices (like [[Cisco Linksys PAP2|PAP2]], [[Cisco Linksys PAP2T|PAP2T]] and [[Cisco SPA2100 Phone Adapter|SPA2100]]) should work with VoIP.ms without issue.

'''Voip.ms recommended dial plan''':

(911S0|310xxxx|<:1'''555'''>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

'''Note''': For 7 digit dialing, replace 555 by the area code of your choice.

This guide has been created to help you learn about Dial Plans. They can be customized according to your preferences. Please note that customizing your dial plan is entirely optional.
 

__TOC__

== What is a Dial Plan? ==

The dial plan is a string of characters that determine how entered phone digits are interpreted and transmitted by your ATA device. It also determines whether to accept, or reject, a call. A dial plan thus facilitates dialing, and also the blocking, of certain types of calls, such as long distance or international.

'''Please Note''': International calls can also be blocked within your VoIP.ms account.

== Digit Sequence ==

A dial plan contains a series of digit sequences, separated by the | character, entirely enclosed within parentheses. Each time a phone button is pressed, your ATA device will attempt to match the digit sequence in your dial plan.

{| class="wikitable"
|-
! Digit Sequence
! Function
|-
| style="text-align:center;"|0 1 2 3 4 5 6 7 8 9 * #
| You can use any of these characters to represent a pressed phone digit.
|-
| style="text-align:center;"| x
| Any phone digit.
|-
| style="text-align:center;"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *.
|-
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. For example, '''01.''' allows the user to enter 0, 01, 011 and so on.
|-
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The ''dialed'' digits can be zero or more characters. For example with this sequence '''<:1555>xxxxxxx''' if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press 6782345, the system transmits 15556782345.
|-
| style="text-align:center;"|, (comma)
| This can be used between digits to play an “outside line” dial tone after a user-entered sequence. For example, with this sequence '''9, 1x.''' an “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1
|-
| style="text-align:center;"|! (exclamation point)
| You can use this character to prohibit a dial sequence. For example with the sequence '''1900xxxxxxx!''' the system reject any sequence that starts with 1900.
|-
| style="text-align:center;"|S0 or L0
| Overrides the Short or Long inter-digit timer to 0 seconds.
|-
| style="text-align:center;"|P# (where # is the duration of the pause in seconds)
| Pauses # seconds.
|}

== Examples ==

Some examples of dial plan digit sequences:

* To dial any international number without using the 011 prefix.
<:011> [2-9]xxxxxxxx.

You can also accomplish this if you set the Dialing Mode to E164 in your [[Account Settings]]

* To block a call to a specific area code (replace 555 with the area code you want)
<:1> 555 xxxxxxx !

* The next sequence, allows you to dial your [[Phone book]] entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520
<:*75>xx< # : >

== See also ==
ATA's
* [[Cisco SPA2100 Phone Adapter]]
* [[Cisco SPA2102 Phone Adapter with Router]]
* [[Cisco Linksys PAP2]]
* [[Cisco Linksys PAP2T]]

IP Phones
* [[Cisco SPA504G Phone|Cisco SPA300/500-series 'phones]]
* [[Cisco Linksys SPA942 NA]]

[[Category:Guides]]

== References ==
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters]
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones]

[[Category: SIP devices]]

Dial Plan for Linksys ATAs

2013-04-17T19:34:03Z

Bills:

The basic dial plan provided in the configuration samples for the Linksys ATA devices (like [[Cisco Linksys PAP2|PAP2]], [[Cisco Linksys PAP2T|PAP2T]] and [[Cisco SPA2100 Phone Adapter|SPA2100]]), should work with VoIP.ms without issue.

'''Voip.ms recommended dial plan''':

(911S0|310xxxx|<:1'''555'''>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

'''Note''': For 7 digit dialing, replace 555 by the area code of your choice.

This guide has been created to help you learn about Dial Plans. They can be customized according to your preferences. Please note that customizing your dial plan is entirely optional.
 

__TOC__

== What is a Dial Plan? ==

The dial plan is a string of characters that determine how entered phone digits are interpreted and transmitted by your ATA device. It also determines whether to accept or reject a call. A dial plan thus facilitates dialing, and also the blocking, of certain types of calls, such as long distance or international.

'''Note''': Please note: International calls can also be blocked within your VoIP.ms account.

== Digit Sequence ==

A dial plan contains a series of digit sequences, separated by the | character and the entire set of sequences is enclosed within parentheses. Each time you press a key in your keypad your ATA device is going to try matching that key with each digit sequence in your dial plan.

{| class="wikitable"
|-
! Digit Sequence
! Function
|-
| style="text-align:center;"|0 1 2 3 4 5 6 7 8 9 * #
| You can use any of these characters to represent a key pressed in your keypad.
|-
| style="text-align:center;"| x
| This represent any character on the phone.
|-
| style="text-align:center;"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *.
|-
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. For example, '''01.''' allows the user to enter 0, 01, 011 and so on.
|-
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The ''dialed'' digits can be zero or more characters. For example with this sequence '''<:1555>xxxxxxx''' if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press 6782345, the system transmits 15556782345.
|-
| style="text-align:center;"|, (comma)
| This can be used between digits to play an “outside line” dial tone after a user-entered sequence. For example, with this sequence '''9, 1x.''' an “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1
|-
| style="text-align:center;"|! (exclamation point)
| You can use this character to prohibit a dial sequence. For example with the sequence '''1900xxxxxxx!''' the system reject any sequence that starts with 1900.
|-
| style="text-align:center;"|S0 or L0
| Overrides the Short or Long inter-digit timer to 0 seconds.
|-
| style="text-align:center;"|P# (where # is the duration of the pause in seconds)
| Pauses # seconds.
|}

== Examples ==

Some examples of dial plan digit sequences:

* To dial any international number without using the 011 prefix.
<:011> [2-9]xxxxxxxx.

You can also accomplish this if you set the Dialing Mode to E164 in your [[Account Settings]]

* To block a call to a specific area code (replace 555 with the area code you want)
<:1> 555 xxxxxxx !

* The next sequence, allows you to dial your [[Phone book]] entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520
<:*75>xx< # : >

== See also ==
ATA's
* [[Cisco SPA2100 Phone Adapter]]
* [[Cisco SPA2102 Phone Adapter with Router]]
* [[Cisco Linksys PAP2]]
* [[Cisco Linksys PAP2T]]

IP Phones
* [[Cisco SPA504G Phone|Cisco SPA300/500-series 'phones]]
* [[Cisco Linksys SPA942 NA]]

[[Category:Guides]]

== References ==
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters]
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones]

[[Category: SIP devices]]

Dial Plan for Linksys ATAs

2013-04-17T19:24:12Z

Bills: Update recent changes.

The basic dial plan provided in the configuration samples for the Linksys ATA devices (like [[Cisco Linksys PAP2|PAP2]], [[Cisco Linksys PAP2T|PAP2T]] and [[Cisco SPA2100 Phone Adapter|SPA2100]]), should work without any issue with VoIP.ms.

'''Voip.ms recommended dial plan''':

(911S0|310xxxx|<:1'''555'''>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

'''Note''': Replace 555 by the area code of your choice. If you want to have the 7 dial for your area code.

This guide has been created in order to help you learn more about the Dial Plan and also you can customize it according to your preferences. Please note that customizing your dial plan is optional.

__TOC__

== What is a Dial Plan? ==

The dial plan is a string of characters that determine how the digits input in your keypad are interpreted and transmitted by your ATA device. It also determines whether or not to accept or reject a call. A dial plan thus facilitates dialing, and also the blocking of certain types of calls, such as long distance or international.

'''Note''': Please notice, that you can also block the international calls in your VoIP.ms account.

== Digit Sequence ==

A dial plan contains a series of digit sequences, separated by the | character and the entire set of sequences is enclosed within parentheses. Each time you press a key in your keypad your ATA device is going to try matching that key with each digit sequence in your dial plan.

{| class="wikitable"
|-
! Digit Sequence
! Function
|-
| style="text-align:center;"|0 1 2 3 4 5 6 7 8 9 * #
| You can use any of these characters to represent a key pressed in your keypad.
|-
| style="text-align:center;"| x
| This represent any character on the phone.
|-
| style="text-align:center;"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *.
|-
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. For example, '''01.''' allows the user to enter 0, 01, 011 and so on.
|-
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The ''dialed'' digits can be zero or more characters. For example with this sequence '''<:1555>xxxxxxx''' if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press 6782345, the system transmits 15556782345.
|-
| style="text-align:center;"|, (comma)
| This can be used between digits to play an “outside line” dial tone after a user-entered sequence. For example, with this sequence '''9, 1x.''' an “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1
|-
| style="text-align:center;"|! (exclamation point)
| You can use this character to prohibit a dial sequence. For example with the sequence '''1900xxxxxxx!''' the system reject any sequence that starts with 1900.
|-
| style="text-align:center;"|S0 or L0
| Overrides the Short or Long inter-digit timer 0 seconds.
|-
| style="text-align:center;"|P# (where # is the duration of the pause in seconds)
| Pauses # seconds.
|}

== Examples ==

Some examples of dial plan digit sequences:

* To dial any international number without using the 011 prefix.
<:011> [2-9]xxxxxxxx.

You can also accomplish this if you set the Dialing Mode to E164 in your [[Account Settings]]

* To block a call to a specific area code (replace 555 with the area code you want)
<:1> 555 xxxxxxx !

* The next sequence, allows you to dial your [[Phone book]] entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520
<:*75>xx< # : >

== See also ==
ATA's
* [[Cisco SPA2100 Phone Adapter]]
* [[Cisco SPA2102 Phone Adapter with Router]]
* [[Cisco Linksys PAP2]]
* [[Cisco Linksys PAP2T]]

IP Phones
* [[Cisco SPA504G Phone|Cisco SPA300/500-series 'phones]]
* [[Cisco Linksys SPA942 NA]]

[[Category:Guides]]

== References ==
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters]
* [http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones]

[[Category: SIP devices]]

Talk:Dial Plan for Linksys ATAs

2013-01-11T18:31:05Z

Bills: Minor Edit.

"Note: Replace 555 by the area code of your choice. If you want to have the 7 dial for your area code." - it would be useful for those of us like in the 519 area code where all dialing is 10 digits, to see a version of the recommendation without the 7 digit part(s).

"<:1> 555 xxxxxxx ! " - should the space before the '!' be removed? If not (it doesn't matter), it would be useful to note that.

Talk:Dial Plan for Linksys ATAs

2013-01-11T18:07:06Z

Bills: 10-digit dialling only, equivalent recomendation?

Talk:CallerID Filtering

2013-01-10T19:48:14Z

Bills: Additional details.

Talk:CallerID Filtering

2013-01-10T18:58:45Z

Bills: Steps to enable filters; multiple filters?

Category:VoIP

2013-01-10T18:35:58Z

Bills: Create a placeholder page for the referenced but non-existent VoIP page

VoIP placeholder page.

Talk:Ring Groups

2013-01-10T18:28:09Z

Bills: Call to main rings all subaccounts?

It would be useful to know here, and on the subaccounts article, if a call to one's main account automatically rings all subaccounts, or if a ring group must be set up for that.

Talk:Voicemail

2013-01-10T18:20:13Z

Bills: Add: instruction instructions; no need for phone recording.

If there is a digit that can be pressed, if 'play instructions before beep' is set to 'no', to play those instructions, it would be useful to note that here. [Later in the article it notes 'press 0 (zero) for options' - if that is true for hearing instructions, it would be useful to so note.]

I would guess that the expected use case is to have a caller hear a message, then be sent to voice mail. As I read this article, 'voicemail' sets up voice mail, once reached. No mention of a message is contained in either 'Assigning your Voicemail' sections. If appropriate, could a setup case be described wherein the call (upon no answer, for example) is routed to a phone recording (saying, for example, "Please leave a message after the beep."), which is in turn routed to voice mail. If this is not the correct setup procedure, mentioning the correct procedure would be useful.

- reading further, I now see that the message prompts are recorded via the voice mail system itself. It would be worth noting that earlier in the article, preventing the question above.

Voicemail

2013-01-10T18:18:43Z

Bills: /* Navigate the Voicemail Menu */ - minor edit.

VoIP.ms has an advanced Voicemail feature that is free to use, and you also have the option to forward your messages to your email address as an attachment.

In order to use the Voicemail feature with Voip.ms you will have to create a Voicemail entry and then assign your entry to one of your DIDs or Accounts.

Please note that Voicemail system is not centralized. It is independent and per server. For Example, If your DID is currently on the Los Angeles POP (Server) and you dial *97 while registered on New York, you will not access the same Mailbox. The same issue can occur if you change the POP (server) of your DID to another server.

If you would like to migrate your Voicemail messages and unavailable messages to another server, please send a request to the VoIP.ms technical support team to support@voip.ms including Voicemail number, Current server and the server you would like to transfer your setting to.

'''The current voicemail messages limit on a single Mailbox is 100'''

== Creating a Voicemail entry: ==

From the Customer Portal refer to DID Numbers -> Voicemail
You will see the following screen.

[[File:Voicemail_entry.JPG|900px]]

You will be prompt with the following information:

'''Name:''' This can be use as a note or description to easily identified your mail boxes.

'''Mailbox ID:''' This will be used as a unique identifier for your mail box. The minimum is adding one digit to up to five digits, for example you can set 1xxxx-1 or 1xxxx-54321.

'''Password:''' The password is used to enter your mail box options such as listen your messages or record your greeting.

'''Skip Password Prompt:''' If set to Yes, when dialing *97 from an account associated to this mailbox, it will skip the password prompt and login directly.

'''Email:''' If an email address is entered here, the Mailbox system will send an Email notification every time you receive a new message. For the moment you can only set 1 email address, however you can optionally configure an Email forward between your email accounts as a work around.

'''Attached message to email:''' If set to YES, the Mailbox will attached a .WAV file containing the new message every time it sends an Email notification.

'''Delete Messages after Emailed:''' If set to YES, the Mailbox will delete the new message automatically after sending the Email notification with attachment.

'''Say Time Envelope:''' If set to YES, when checking your messages you will hear the date and time when the message was received.

'''Time Zone:''' The time envelope will use this time zone to provide the correct date and time of the messages reception.

'''Say Caller ID:''' If set to YES, when checking your messages you will hear the Caller ID of the message sender.

'''Play Instructions Before Beep:''' If set to YES, the caller will hear instructions on how to leave a message to your Mailbox before the beep sound.

'''Voicemail Menu Language:''' This sets the language you and the caller will hear when instructions or menus are played by the voicemail.

== Assigning your Voicemail to your DID==

After you have created your Voice Mail entry, you can assign it to any of your DIDs from your main portal. Please refer to DID Numbers -> [[Manage DID]] -> Edit DID -> Voicemail. Also under the same screen you can set the Dial Time Out (The maximum amount of time a call to your DID can stay in "Ringing State" before we cancel the call to no answer). Please note that 30s equals to 6 rings.

== Assigning your Voicemail to your Account==

If you would like to assign a Voicemail entry to your Main Account, please from your main portal refer to Main Menu -> [[Account Settings]] -> General -> Voicemail Associated to the Main Account.

[[File:Mainvoicemail.jpg]]

If you need to assign a Voicemail to a specific subaccount, you need to go to the [[Sub Accounts]] Edit page, following the route, Subaccounts >> Manage Subaccounts >> Edit, from the menu tabs.
You will see at the bottom of the page the "Internal Extension Voicemail" option. Here you can set it.

[[File:Subvoicemail.jpg]]

== Manage Voicemail ==

[[File:Man voicemail.jpg|800px]]

You can access this screen once you create the voicemail through your Customer Portal>>DID Numbers>>Voicemail page. From here you can change the settings in your voicemail (Modify action), delete the mailbox or delete all the messages in the given mailbox.

Note: However you cannot change the Mailbox ID or the default access code.

== Voicemail Access Codes: ==

You can access your voicemail with any device/system connected directly with your account or subaccount to VoIP.ms using the codes below:

*<nowiki>*97</nowiki> (Asterisk 97) is used to access directly the Mailbox associated to the account you are dialing from. If you would like to check which mailbox is associated to your account refer to [[Voicemail#Assigning_your_Voicemail_to_your_Account|Assign Voicemail]]

*<nowiki>*98</nowiki> (Asterisk 98) is used to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If for any reason you do not have access to our VoIP network, you can check your Voicemail by just dialing your DID. Once the Voicemail system answer your call, press the asterisk key (*). You will be prompt for Mailbox ID and Password, once logged in to your Voicemail, press 0 (zero) for options. You can also record your greeting and temporary greeting from there.

== Navigate the Voicemail Menu ==

Once you access your voicemail you're going to be prompted with the number of new and/or old messages you have in the mailbox. Here's the list of options you have with the voicemail system of VoIP.ms

* 1 - Play the first new/old message available in your mailbox.

* 2 - Change folders. This option allows you to change to another folder in order to hear the messages you have stored in that folder. At the moment it's not possible to change the name of the folders.
0 - New Messages
1 - Old Messages
2 - Work
3 - Family
4 - Friends
# - Cancel

* 0 - Voicemail options. In here you can change your greetings and record your name, also you can change the password for your voicemail.
1 - Unavailable message
2 - Busy message
3 - Name.
4 - Temporary message
5 - Change Password
* - Return to the Main Menu

'''Note:''' The Unavailable and Busy messages are not currently working with the voicemail system of VoIP.ms.
Use the temporary greeting, which overrides these standard greetings.

The follow options are available when you're listening to your messages. However you can use this options inside the main menu, and they are apply to the first new/old message.

* 3 - Advanced Options
1 - Send a reply. Currently not available.
2 - Message Envelope. Speak the date and time at which the message was received.
* - Return to the Main menu.

* 4 - Play the previous message.

* 5 - Repeat the message.

* 6 - Play the next message.

* 7 - Delete the current message, without confirmation.

* 8 - Forward message to another user. Prompt for the internal extension number.
1 - Prepend the message with a recording.
2 - Send the message without a prepending message.
* - Return to the Main Menu

[[category:guides]]

Voicemail

2013-01-10T18:16:28Z

Bills: /* Navigate the Voicemail Menu */ - minor edit.

VoIP.ms has an advanced Voicemail feature that is free to use, and you also have the option to forward your messages to your email address as an attachment.

In order to use the Voicemail feature with Voip.ms you will have to create a Voicemail entry and then assign your entry to one of your DIDs or Accounts.

Please note that Voicemail system is not centralized. It is independent and per server. For Example, If your DID is currently on the Los Angeles POP (Server) and you dial *97 while registered on New York, you will not access the same Mailbox. The same issue can occur if you change the POP (server) of your DID to another server.

If you would like to migrate your Voicemail messages and unavailable messages to another server, please send a request to the VoIP.ms technical support team to support@voip.ms including Voicemail number, Current server and the server you would like to transfer your setting to.

'''The current voicemail messages limit on a single Mailbox is 100'''

== Creating a Voicemail entry: ==

From the Customer Portal refer to DID Numbers -> Voicemail
You will see the following screen.

[[File:Voicemail_entry.JPG|900px]]

You will be prompt with the following information:

'''Name:''' This can be use as a note or description to easily identified your mail boxes.

'''Mailbox ID:''' This will be used as a unique identifier for your mail box. The minimum is adding one digit to up to five digits, for example you can set 1xxxx-1 or 1xxxx-54321.

'''Password:''' The password is used to enter your mail box options such as listen your messages or record your greeting.

'''Skip Password Prompt:''' If set to Yes, when dialing *97 from an account associated to this mailbox, it will skip the password prompt and login directly.

'''Email:''' If an email address is entered here, the Mailbox system will send an Email notification every time you receive a new message. For the moment you can only set 1 email address, however you can optionally configure an Email forward between your email accounts as a work around.

'''Attached message to email:''' If set to YES, the Mailbox will attached a .WAV file containing the new message every time it sends an Email notification.

'''Delete Messages after Emailed:''' If set to YES, the Mailbox will delete the new message automatically after sending the Email notification with attachment.

'''Say Time Envelope:''' If set to YES, when checking your messages you will hear the date and time when the message was received.

'''Time Zone:''' The time envelope will use this time zone to provide the correct date and time of the messages reception.

'''Say Caller ID:''' If set to YES, when checking your messages you will hear the Caller ID of the message sender.

'''Play Instructions Before Beep:''' If set to YES, the caller will hear instructions on how to leave a message to your Mailbox before the beep sound.

'''Voicemail Menu Language:''' This sets the language you and the caller will hear when instructions or menus are played by the voicemail.

== Assigning your Voicemail to your DID==

After you have created your Voice Mail entry, you can assign it to any of your DIDs from your main portal. Please refer to DID Numbers -> [[Manage DID]] -> Edit DID -> Voicemail. Also under the same screen you can set the Dial Time Out (The maximum amount of time a call to your DID can stay in "Ringing State" before we cancel the call to no answer). Please note that 30s equals to 6 rings.

== Assigning your Voicemail to your Account==

If you would like to assign a Voicemail entry to your Main Account, please from your main portal refer to Main Menu -> [[Account Settings]] -> General -> Voicemail Associated to the Main Account.

[[File:Mainvoicemail.jpg]]

If you need to assign a Voicemail to a specific subaccount, you need to go to the [[Sub Accounts]] Edit page, following the route, Subaccounts >> Manage Subaccounts >> Edit, from the menu tabs.
You will see at the bottom of the page the "Internal Extension Voicemail" option. Here you can set it.

[[File:Subvoicemail.jpg]]

== Manage Voicemail ==

[[File:Man voicemail.jpg|800px]]

You can access this screen once you create the voicemail through your Customer Portal>>DID Numbers>>Voicemail page. From here you can change the settings in your voicemail (Modify action), delete the mailbox or delete all the messages in the given mailbox.

Note: However you cannot change the Mailbox ID or the default access code.

== Voicemail Access Codes: ==

You can access your voicemail with any device/system connected directly with your account or subaccount to VoIP.ms using the codes below:

*<nowiki>*97</nowiki> (Asterisk 97) is used to access directly the Mailbox associated to the account you are dialing from. If you would like to check which mailbox is associated to your account refer to [[Voicemail#Assigning_your_Voicemail_to_your_Account|Assign Voicemail]]

*<nowiki>*98</nowiki> (Asterisk 98) is used to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If for any reason you do not have access to our VoIP network, you can check your Voicemail by just dialing your DID. Once the Voicemail system answer your call, press the asterisk key (*). You will be prompt for Mailbox ID and Password, once logged in to your Voicemail, press 0 (zero) for options. You can also record your greeting and temporary greeting from there.

== Navigate the Voicemail Menu ==

Once you access your voicemail you're going to be prompted with the number of new and/or old messages you have in the mailbox. Here's the list of options you have with the voicemail system of VoIP.ms

* 1 - Play the first new/old message available in your mailbox.

* 2 - Change folders. This option allows you to change to another folder in order to hear the messages you have stored in that folder. At the moment it's not possible to change the name of the folders.
0 - New Messages
1 - Old Messages
2 - Work
3 - Family
4 - Friends
# - Cancel

* 0 - Voicemail options. In here you can change your greetings and record your name, also you can change the password for your voicemail.
1 - Unavailable message
2 - Busy message
3 - Name.
4 - Temporary message
5 - Change Password
* - Return to the Main Menu

'''Note:''' The Unavailable and Busy messages are not currently working with the voicemail system of VoIP.ms.
Use the temporary greeting, which overrides these standard greetings.

The follow options are available when you're listening to your messages. However you can use this options inside the main menu, and they are apply to the first new/old message.

* 3 - Advanced Options
1 - Send a reply. Currently not available.
2 - Message Envelope. Speak the date and time at which the message was received.
* - Return to the Main menu.

* 4 - Play the previous message.

* 5 - Repeat the message.

* 6 - Play the next message.

* 7 - Delete the current message, without confirmation.

* 8 - Forward message to another user. Prompt for the internal extension number.
1 - Prepend the message with a recording.
2 - Send the message without any prepend message.
* - Return to the Main Menu

[[category:guides]]

Talk:DISA

2013-01-10T17:53:24Z

Bills: DISA Rate example - call to toll free out to local / long distance destination.

DISA

2013-01-10T17:44:20Z

Bills: /* How calls will be billed using DISA? */ - minor edits.

'''What is DISA ?'''
----
Direct Inward System Access ( DISA ) allows you to use our system for placing outgoing calls, even if you are not close to any device where you are registering your account or sub account. In this case you just would need to dial to your DID number and to provide a 4 digits PIN number, then you can dial out to any number in the world under our termination rates.

__TOC__

== Setup DISA ==

=== How you can create a DISA entry ===
----

1) Please login on the customer portal, then go to DID numbers -> DISA. There you will see this:

[[File:DISA1.JPG]]

2) Once you click on button "Add DISA" you will be able to see this:

[[File:DISA2.JPG]]

- '''DISA Name''': Here you can enter the name which best describes this entry.

- '''DISA PIN''': It is a 4 digits number that you will need to provide in order to get dial tone.

- '''Digit Timeout''': The maximum amount of time( in seconds ) our system will wait between digits.

- '''CallerID Number Override''': Here you can set the number you want that people receive when you dial out from DISA.

3) Once you enter all this information, you would need to click on button "Save" to see DISA entry in this way:

[[File:DISA3.JPG]]

=== Where you can use a DISA entry ===
----
You can use a DISA entry for:

1) '''Routing your DID number.'''

2) '''Pointing one extension of a Digital Receptionist ( [[Digital Receptionist (IVR)|IVR]] ).'''

3) '''Setting as destination into a [[Time Conditions|Time Condition]], or'''

4) '''Routing a call according what you set on the [[CallerID Filtering|CallerID Filtering]].'''

IMPORTANT NOTE : From DISA you can dial out to any number, includinga Speed Dial Entry you have created in your [[Phone book|Phone
book]]. For example: *75XX and then press the pound key (#) or simply wait for the duration of the "Digit Time-out", and our system will
place the call.

=== How calls will be billed using DISA? ===
----
The cost of these calls will depend of the billing plan of your DID number (per minute or flat rate), the destination of the outgoing call and the time of the call. For example:

1) If you have a DID number under per minute plan with a rate for incoming of $0.0100 (per minute) and this number is routed to DISA, when someone calls and places an outgoing call for 1 minute to a Canadian number the cost will be:

A -> $0.0100 + $0.0052 = $0.0152 [value routing], or

B -> $0.0100 + $0.0125 = $0.0225 [premium routing].

2) If you have a DID number under flat rate plan with a rate for incoming of $0.00 (per minute) and this number is routed to DISA, when someone calls and places an outgoing call for 1 minute to a Canadian number the cost will be:

C -> $0.00 + $0.0052 = $0.0052 [value routing], or

D -> $0.00 + $0.0125 = $0.0125 [premium routing].

[[category:guides]]

DISA

2013-01-10T17:37:37Z

Bills: /* Where you can use a DISA entry */ - minor edit.

'''What is DISA ?'''
----
Direct Inward System Access ( DISA ) allows you to use our system for placing outgoing calls, even if you are not close to any device where you are registering your account or sub account. In this case you just would need to dial to your DID number and to provide a 4 digits PIN number, then you can dial out to any number in the world under our termination rates.

__TOC__

== Setup DISA ==

=== How you can create a DISA entry ===
----

1) Please login on the customer portal, then go to DID numbers -> DISA. There you will see this:

[[File:DISA1.JPG]]

2) Once you click on button "Add DISA" you will be able to see this:

[[File:DISA2.JPG]]

- '''DISA Name''': Here you can enter the name which best describes this entry.

- '''DISA PIN''': It is a 4 digits number that you will need to provide in order to get dial tone.

- '''Digit Timeout''': The maximum amount of time( in seconds ) our system will wait between digits.

- '''CallerID Number Override''': Here you can set the number you want that people receive when you dial out from DISA.

3) Once you enter all this information, you would need to click on button "Save" to see DISA entry in this way:

[[File:DISA3.JPG]]

=== Where you can use a DISA entry ===
----
You can use a DISA entry for:

1) '''Routing your DID number.'''

2) '''Pointing one extension of a Digital Receptionist ( [[Digital Receptionist (IVR)|IVR]] ).'''

3) '''Setting as destination into a [[Time Conditions|Time Condition]], or'''

4) '''Routing a call according what you set on the [[CallerID Filtering|CallerID Filtering]].'''

IMPORTANT NOTE : From DISA you can dial out to any number, includinga Speed Dial Entry you have created in your [[Phone book|Phone
book]]. For example: *75XX and then press the pound key (#) or simply wait for the duration of the "Digit Time-out", and our system will
place the call.

=== How calls will be billed using DISA? ===
----
The cost of these calls will depend of the billing plan of your DID number( per minute or flat rate ), the destination of the outgoing call and the time of the call. Per example:

1)If you have a DID number under per minute plan with a rate for incoming of $0.0100( per minute ) and this number is routing to DISA. When someone calls and place an outgoing call of 1 minute to a Canadian number the cost will be:

A ->$0.0100 + $0.0052 = $0.0152 [if customer uses value route], or

B ->$0.0100 + $0.0125 = $0.0225 [if customer uses premium route].

2)If you have a DID number under flat rate plan with a rate for incoming of $0.00( per minute ) and this number is routing to DISA. When someone calls and place an outgoing call of 1 minute to a Canadian number the cost will be:

C ->$0.00 + $0.0052 = $0.0052 [if customer uses value route], or

D ->$0.00 + $0.0125 = $0.0125 [if customer uses premium route].

[[category:guides]]

Recordings

2013-01-10T17:23:30Z

Bills: Minor edits.

'''How to use your own Recordings?'''
----

VoIP.ms allows you to upload an audio file and use in the different options we have under DID numbers menu. It can be used with several of the other features of our system such as [[Digital Receptionist (IVR)|Digital receptionist]], [[Calling Queues|Calling queues]], and others.

The sound file should be a Windows .WAV sound file (extension .wav) with a format: PCM 8kHz 16 bits Mono.

__TOC__

== Setup Recording ==

=== How you can upload a Recording ===
----
To upload a recording, you need to login on the customer portal, then you have to go to DID numbers -> Recordings. There you will see this:

[[File:Recording2.JPG]]

- '''Name:''' you can set the name you want to identify this recording

- '''File:''' here you need to select the recording you want to upload on your account.

- '''Upload:''' click on this to start uploading procedure.

IMPORTANT NOTE: When you upload a new recording the system can take up to 60 seconds to propagate this recording to all VoIp servers. It will not be playable until this process is complete.

Once you upload a recording you will see:

[[File:Recording3.JPG]]

=== Recording Options ===
----

Once a recording is in your account you can execute the following actions:

- '''Test Dial Code''': Dial this code to hear the uploaded recording and confirm everything is correct.

IMPORTANT NOTE : The system will attempt to convert your .wav file into the required format once the upload is complete, however, the result can not be guaranteed if the file is not in the proper format.

- '''Download''': Use this option if you want to download this recording from your account.

- '''Re-upload''': Use this option if you want to upload again this recording in your account.

- '''Delete''': Use this option if you want to delete this recording from your account when you do not longer need it..

=== How you can use a Recording ===
----

From our system your recordings can be used in different features, such as:

- '''[[Digital Receptionist (IVR)|Digital receptionist]]''' on customer portal -> DID numbers.

- '''[[Calling Queues|Calling queues]]''' on customer portal -> DID numbers.

- '''To route a DID number''' on customer portal -> DID numbers -> Manage DID -> Select DID -> edit DID -> Routing

- '''To route failover options''' on customer portal -> DID numbers -> Manage DID -> Select DID -> edit DID -> Failover

[[category:guides]]

Caller ID

2013-01-10T15:45:01Z

Bills: /* Incoming Caller ID number and name */ - minor corrections

Caller ID is a telephone service that transmits the calling party´s number to the called party´s telephone . When available the Caller ID number can be complemented with the caller ID name (description e.g. John Smith)

If you are placing outgoing calls, most likely you will need to pass a Caller ID to ensure proper termination of your calls, along with being able to reach toll free numbers properly.

There are two types of caller ID and its important to understand they are 2 different things: these are Caller ID Name and Caller ID Number.

Please note that the caller ID is only guaranteed while using premium route, and only for US48 and Canada calls, even though you may find your caller ID working on Value route for some routes.

We will check on incoming Caller ID (Caller ID from people calling you) later on this article.

__TOC__

== Outgoing Caller ID number ==

This is the most common Caller ID type you will be passing. If you will be using a more complex system capable of passing its own caller ID Like a [[Welcome#PBX|PBX]] you most likely would like to set this from the Caller ID field from the trunk, or from any of its extensions.

If you are using devices like [[Welcome#Devices|Telephone Adapters]], [[Welcome#Devices|IP phones]] or [[Welcome#Softphones|softphones]], the caller ID number is just one more setting from your voip.ms account and you require to set this from the customer portal.

To set the Caller ID number for your Main account, you need to enter the "[[Account Settings]]" from the "Main Menu" menu tab, and then head for the General Tab.

[[File:CIDmain.jpg|Main account Caller ID]]

If you need to set the Caller ID number for your subaccount, you can set this at the moment of creating it, or later simply by going to the Edit page of the [[Sub Accounts|subaccount]]. It will work the same way.

'''It is strongly suggested to set a 10 digits valid Caller ID number, to ensure proper termination. The Caller ID field from the portal only supports numerical characters.'''
'''Anonymous and Toll free caller IDs are not recommended as they could cause troubles to connect, specially when Calling Toll free numbers.'''

== Outgoing Caller ID name ==

The caller ID name is an additional information you can pass along with your Caller ID number. This will also be received on the callee's end and it could be your name or you Business.

For example: ''"John Smith"<9145551234>''

The sample above is a Caller ID, that includes both Caller ID name and Caller ID number, commonly abbreviated as CID and CNAM among other variations.

Is not possible to set any Caller ID name from the voip.ms portal.

If you will be making calls to Canadian numbers, you can simply pass the Caller ID name from your device or system (if it supports it, most [[Softphones]] do). You will need to check for a field on the interface from the device to enter this setting, and in case you are using a more advanced system, get assistance to set the outgoing caller ID name set up.

The Caller ID name on US however works different, this is controlled by a national CNAM database, with records of numbers and names matching each number.
When you make a call to a US number, you will send a caller ID number, and the system will check on the CNAM database for a name matching the same Caller ID number you passed, in order to display both name and number to final phone.

CNAM is only available for some USA numbers. In order to update your Caller ID on the CNAM database for your US calls, there is a process to follow which has a cost of $10 USD (one time only).
Please contact voip.ms support to get further details on what information you need to submit and to confirm if your local US DID is available for CNAM update.

'''CNAM update is only available for some Local US DIDs. Toll frees can not have their Caller ID name updated'''.

== Incoming Caller ID number and name ==

You will receive the Caller ID number and Caller ID name that the voip.ms server receives from the caller, this is exactly what will be sent to you on Incoming calls.
You can always check what Caller ID number voip.ms receives, by going into your [[Call Detail Records]] to check the incoming calls.

The incoming caller ID name, works almost the same way, except that this is an optional setting that you need to enable from per DID number on the [[Manage DID|DID settings]] page.
This option is called "Caller ID Lookup". When enabled, the system will perform a query on the LIBD/CNAM Database, for callers with Canadian or US CID number, in order to find a name matching that CID number.
The system then will display the result of this query in the Caller ID name portion of the '''Caller ID''', leading to a "Caller ID name"<5551231234> when people call your number.

----

'''On a side note, outgoing Caller ID is not guaranteed on calls to Canadian cellular numbers, even when using the premium route. This is due to the way Canadian carriers work - they sometimes pass a random Caller ID that they have on record, changing the original.'''
'''This is out of our control as it is the way Canadian carriers handle calls to cellular numbers.'''

[[category:guides]]

Calling Queues

2013-01-10T15:32:43Z

Bills: Minor corrections.

__TOC__

== Concept ==
If you want a solution to manage your incoming calls, and have your customer(s) wait on the line while an agent picks up the call, you need to create a Calling Queue entry, this will permit you to have many calls in hold, queued calls in First In, First Out order until agents become available.

Queues consist of:

*Incoming calls being placed in the queue
*Members that answer the queue (extensions or users that login as agents)
*A strategy for how to handle the queue and divide calls between members
*Music played while waiting in the queue
*Announcements for members and callers

Agents are the people (or person) that answer call(s) that have been placed into a specific Queue. An agent logs in indicating that s/he is now ready to take calls. Transfers an inbound call to a queue, which is then in turn transfered to an available agent.

The members in the queue can be static or dynamic. The '''Static Members''' are those that are always connected to the queue, on the other hand the '''Dynamic members''' are those that need to log in to the queue in order to take calls.

=== Log in or log out to the queue as Dynamic Member ===
----

If you want get access to your Call queue ,dial *11, at the prompt, dial the queue ID, and password if you set one. e.g. If I want log in to the "call queue 1" I dial *11 >> Option 1 >> Password (optional).

If you want to log out from the call queue, dial *12, at the prompt, dial queue ID. e.g. If I want to log out to the "call queue 1" I dial *12 >> Option 1.

== Configuration ==

Go in to your Customer Portal >> DID Numbers >> Calling Queues

[[File:Screenshot (11h 41m 41s).jpg|400px]]

And then go to "Create New Call Queue"

[[File:Screenshot 2.jpg]]

Now you are going to start a new configuration:

===Queue Information===

[[File:Screenshot3.jpg|400px]]

*'''Queue Number''': The number of our Queue, e.g I have my company and I want select the Queue 1 to my sales department, my Queue 2 to my support department. In this way you are sure that your agents receive the calls properly.

*'''Queue Name''': Enter the name of you Call Queue 1, e.g. "Sales Team".

*'''Queue Language''': The language of system announcements. Here, "English" .

*'''Queue Password''': An optional setting - you can preset a password to access to this queue and be sure that only the people authorized for this queue enter it.

*'''Caller ID Prefix''': Optional: You can optionally prefix the [[Caller ID]].

*'''Join Announcement''': If you have a recording for your queue you can set it here. This recording plays when a member enters the queue. e.g. In my company, I have a queue for my sales department, while my customers wait, they hear a recording of all our products and discounts.

*'''Priority / Weight''': Weight of queue, compared to other queues. If an agent is logged in to more than 1 queue, the higher weighted queue calls that agent.

===Queue Options===
----

[[File:Screenshot 4.jpg|400px]]

*'''Agent Announcement''': Optionally, you can set a recording to be played to the agent. The caller will be connected to the agent immediately after the announcement.

*'''Report Hold time to agent''': If you wish to report the caller's hold time to the agent, set this to yes.

*'''Member Delay''': If you wish to have a delay before the agent is connected to the caller, set this to the number of seconds to delay.

*'''Maximum Wait Time''': The maximum time that a caller can wait in queue before being sent to the "Failover" Destination.

*'''Join when empty''':

-'''Yes:''' Callers can join a queue with no members or only unavailable members.
-'''Strict:''' Callers cannot join a queue with no members or only unavailable members.
-'''No:''' Callers cannot join a queue with no members.

*'''Leave when empty''':

-'''Yes:''' Callers are sent to the failover when there are no members.
-'''Strict:''' Callers are sent to failover if there are members but non of theme is available.
-'''No:''' Callers will remain in the queue even if there are no members.

*'''Ring Strategy'''

Calls are distributed among the members handling a queue with one of several strategies, defined in queues.conf

-'''Ringall:''': ring all available channels until one answers.
-'''Leastrecent:''': ring interface which was least recently called by this queue
-'''Fewestcalls:''': ring the one with fewest completed calls from this queue
-'''Random:''': ring random interface
-'''Round Robin Memory:''': round robin with memory, remember where we left off last ring pass

*'''Ring in-use''': This setting let you avoid that the call goes to an agent which device is currently in use.
'''Note:''' Currently only a device with the SIP protocol is able to report his status as 'in use'.

*'''Agent Ring Timeout''': Number of seconds in which the call will remain in ringing state, before being considered as timeout.

*'''Retry Timer''': How long the system will wait before trying with all the members again.

*'''Wrap-up Time''': After a successfully call, this setting let you set the amount of minute the system will wait before sending the call to a free agent.

===Announcements and Fail Over===
----

[[File:Screenshot 5.jpg|400px]]

==== '''Periodic Voice Announcements''' ====
----

'''Voice Announcement''': This is an optional setting, you can choose which recording will be played to the callers of this queue.

'''Frequency of announcement''': Select the periodic interval to play the recording to the callers.

==== '''Periodic Hold Position, Estimated Hold-time announcements and Thank you for your patience announcement.''' ====
----

'''Announce Position frequency''': This setting let you know how often to make any periodic announcement. Optional setting.

'''Announce Round Seconds''': Here you can choose if you want to announce the number of seconds or round to minute. If you want to announce seconds, select the amount to round.

'''If Announce position is enabled, do you also want to report estimated hold-time?''': Either yes, no or only once. Hold time will be announced as the estimated time, or less than 2 minutes when appropriate.

'''Thank you for your patience''': This is an optional setting. Your callers will hear "Thank you for your patience", after announcing the Queue Position and Estimated Hold time left.

==== '''Fail Over Destinations''' ====
----

Here you can choose the destination for the follow failover options:

* '''Timeout''': If the call reach the maximum wait time.
* '''FULL''': If the queue reaches the maximum number of callers.
* '''JOINEMPTY''': A call was sended to the queue but the queue had no members (Only works when '''Join when empty''' is set to '''No''')
* '''LEAVEEMPTY''': The last agent was removed form the queue before all calls were handled (Only works when '''Leave when empty''' is set to '''Yes''').
* '''JOINUNAVAIL''': Same as '''JOINEMPTY''', except that there were still queue members, but all were with status unavailable (SIP Phone logged out for example)
* '''LEAVEUNAVAIL''': Same as '''LEAVEEMPTY''',except that there were still queue members, but all were with status unavailable (SIP Phone logged out for example)

We are finished now click on the Save Queue.

== Static Members ==

Static Members are the predefined and permanently assigned members responsible for answering incoming calls to a queue. You can add as many members as you wish to any given queue.

===Advantages===
The advantage of the static members is that your members do not have to login or logout from the queue using the *11 and *12 commands. The only thing your members have to do is register or unregister their accounts from the VoIP.ms servers.

===Disadvantages===
One of disadvantages of the static members is that your members are unable to log out of the queue, they are permanently assigned instead. Other disadvantage is that your members need to use their account or [[Sub Accounts|sub account]] to be in the queue, this could mean that they wouldn't be able to change devices and log into the queue, they would have to use their assigned device.

===Add an Static Member===
----

First you need to access the '''Calling Queue''' page that is under the '''DID Numbers''' menu in your Customer Portal.

[[File:Man callqueue.jpg]]

From there, you need only to click on the link '''Edit Static Members''', that bring you to the next screen:

[[File:Queue stamem.jpg]]

To add a new static member, you need only to fill the follow information:

[[File:Queue addstatmem.jpg]]

'''Description''': Here you can assign a description to easily identify each member of the queue.

'''Account''': Here you select the account or subaccount that is going to be assigned as a member for this queue.

'''Priority''': This value can be equal or greater than zero. Available members with lower priority will get the calls first. You can have more than one member with the same priority.

== How to use your Queue ==

Once you have created a Queue, you can assign it to many DID numbers as you want without needing to create it again. You need to go to your Customer Portal >> DID Numbers >> Manage DID(s). And from there you can route the DID numbers to go directly to the Queue, using an [[Digital Receptionist (IVR)|IVR]], through a [[Time Conditions|time condition]] or even as a failover option for your DID number.

[[category:guides]]