VoIP.ms Wiki - User contributions [en]

OBi 100/110

2014-11-01T16:11:19Z

Mango:

[[File:OBi110-ATA.jpg|none|200px|center]]

 ''The OBi100 is a single phone port ATA adapter that supports SIP VoIP services. The OBi100 is perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.'' 
 
== Manual Configuration Details ==
 
Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''. 
Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
 

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
 
After this, save all changes and you are ready to move on to the actual configuration.
 
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub) 
:'''DigiMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
 

====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
 

Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
 

===Configuring Voice Services===
 
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
 

Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

== Features Star Codes ==
Please check this link to the Star Codes that are available to activate and deactivate some of the features on your device [http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
 

==Known Issues and Resolutions==
 
=== Settings to avoid direct phone calls to your device in the middle of the night ===
 
Some customers have reported receiving calls in the middle of the night coming from "100" or "101" as callerID. These calls are directly to your device and do not pass through our servers, so we cannot filter them. However, we have some suggestions:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Voice Services -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/FAQ#What_are_the_IP_addresses_of_VoIP.ms.C2.B4_servers_.3F Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

 

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
 
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
 

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
 

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

[[Category:Analog Telephone Adapters]]

SIP URI

2013-12-19T16:03:07Z

Mango: /* Creating a new SIP URI */

A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number. The SIP URI resembles an e-mail address and is written in the following format: x@y:port (x=Username, y=host|domain|IP)

: Example: johnsmith@my-uri.com

A general description of SIP addressing is at [[wikipedia:SIP address]]. The addresses, which use the same user@domain... format as e-mail addresses, allow an individual Internet telephony user to be reached directly online without passing via the public switched telephone network or incurring the corresponding tolls.

A SIP address may be used as a destination to which to forward a voip.ms DID number, as a target for an individual speed dial entry (*75xx) in a voip.ms user address book or as a means to transfer incoming calls into your voip.ms extensions or numbers from outside Internet servers.

# One is to send calls to an external SIP URI, via your DID number,
# A second option is to receive calls via SIP URI, we can achieve this using our DID number or an internal extension from a [[Sub Accounts|sub account]].
# The third option is to use a Virtual number.

__TOC__

== Send calls to an external SIP URI address ==

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).

: '''Make sure the other company or provider supports the use of SIP URI'''

=== 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".

[[Image:Forward.jpg]]

; Examples:
: 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.)

''Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call).''

=== Creating a phone book entry ===
A SIP URI may be associated with a [[phone book]] or speed dial entry in the same manner as any other telephone number.

See [[Phone book#Create a Phone Book Entry with a SIP URI]].

[[File:Pb entry sipuri.jpg|800px]]

'''SIP URI''': Here you can either select '''Use Existing''' or '''Create New''' to assign the [[SIP URI]] to your Phone Book entry.

This replaces alphanumeric addresses (such as sip:user@provider.example.org) with numeric abbreviations (such as *7501) which can be easily dialled from [[devices|IP phones]] that only offer a numeric keypad.

=== Codec Negotation ===

By default when you route your incoming calls to an external SIP URI address, the system sends the INVITE allowing all VoIP.ms supported codecs (ulaw, g729a and GSM).
In that case if you want to use a specific codec (from the supported ones) you need to restrict that in your end. For instance, if you are using an Asterisk/PBX System and only wish to use ulaw codec, you will need to make sure to have the following settings in the trunk:

: disallow=all
: allow=ulaw

== Receiving incoming calls from a SIP URI ==

=== Using your DID number ===
You can receive SIP URI calls using the following format {Number}@sip.voip.ms, this can be used with your local US and Canada numbers, so they can be reached from outside.

[[Image:Did.jpg]]

This format of SIP address is used by services such as [[wikipedia:SIP Broker]] as a means to reach voip.ms subscribers.

Another variant, also valid, is to specify the specific voip.ms server on which your DID is registered, ie:
:sip:4166471234@toronto.voip.ms

=== Using your sub account internal extension ===
When you assign an internal extension for a [[Sub Accounts|sub account]], it can also be used as an external SIP URI. For example, if your extension is 2, you could be reached directly via SIP from another network with a URI like this: 1000002@houston.voip.ms

:(Replace houston.voip.ms by the server you are registered to, 100000 by your account ID and the 2 by your internal extension).

Important: no call flow or filtering can be applied to calls make to the external SIP URI. Calls will immediately ring the device registered to this sub-account.

[[Image:Extension.jpg]]

=== Using iNum ===
Any iNum number (from any provider) is a SIP URI; just append @sip.inum.net

For example, iNum 883510009999999 becomes:
:883510009999999@sip.inum.net

=== Using a Virtual number ===

Virtual SIP numbers are similar to standard DID numbers. The major difference is that virtual SIP numbers are not accessible via "PSTN". They can only be reached via "SIP URI" over internet. For example, if you have a DID number with another provider and they support SIP URI Forwarding, you could forward your number to a virtual number at voip.ms just like if it was one of our numbers.

All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits. The final uri will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server.

: Example SIP URI: 11100000123@houston.voip.ms

[[File:Virtualsip.jpg]]

[[category:guides]]

Call quality issues

2012-03-17T00:27:57Z

Mango:

There are different factors that could potentially affect the quality of your calls. There are several types of sound issues and these can be related to different causes. We will try to mention here some suggestions, so we can identify which type of issue we are experiencing and what things we need to check to start a diagnostic by ourselves.

== Reboot your device ==

Even if you can browse without problems using any browser in the computer, and if you think the Internet is working fine, it is possible that something in the network is affecting the calls, the first thing that we always need to test, is to reboot our ATA device and Router, this way we refresh the connection.

== Choose a server ==

A good recommendation is to send a ping to all servers, this way we can verify the latency and pick the best option for us. (This is just a slight introduction, please refer to our article [[Choosing Server]] for more information.

== Softphone test ==

To rule out your ATA Device or PBX as the source of the issues, you can to test with a simple software that can be used for the same, make calls.

'''How to test using [[Softphones]]?'''

*'''Create a sub account''', this way you do not have to mess with the settings on your ATA device, for now.

*Register the softphone using the sub account credentials and make a call, if the issue is the same, the problem can be in our network, if not, then we can start pointing to our device.

*Use an easy to install softphone, '''ZoIPer and X-Lite''' are recommended by the VoIP.ms staff.

== Choppy/Robotic voice ==

=== Network traffic ===

One of the main reasons sound issues may occur is based on the traffic or congestion on the network. First thing to try is check if the issue can be duplicated is making an internal call with the provider, for example using an Echo test application (by dialing 4443) or a [[voicemail]].

'''Some symptoms that can be present because of the lack of bandwidth available:'''

*Audio cutting in and out (choppy).

*Voice sounding robotic, like if you were talking under the water.

*Audio slowing down or speeding up, intermittently during the call.

'''Now, for test if the bandwidth is affecting our calls:'''

*Disconnect all the [[devices]] from the network

*Disable wireless, to make sure no one else is using your internet.

*If your router has QoS, disable it.

*If you were using software to download stuff from Internet (e.g. Torrents) wait a few minutes for this traffic to subside.

After following all these suggestions, use a single device and try to make a call, if the audio quality is fine, you are probably dealing with lack of bandwidth, and for this case the use of QoS is recommended, and make sure the set up is well done.

=== Test codecs ===

Test with all the codecs such as''' g711u, g729 and GSM'''. Sometimes the issues with the audio can be related with the codecs in use, either because the codec we are using is consuming too much bandwidth for our connection or there is a chance also the device we are using is not supporting this codec very well, or it works better with a different one. In any case, this test can also help in the diagnostic.

Check in your [http://wiki.voip.ms/article/Account_Settings Account] or sub account settings, which codec you are allowing, you can test allowing one by one, until you
get the best result. If using codecs such as G.711 you may try with a lower bitrate codec such as G729a or GSM (if they are supported by your device/software/system).

=== Check your ISP ===

After following these suggestions, you still experience sound issues? You may consider to contact your ISP (Internet provider) just to confirm the issue is not related with them.

== Tones during calls ==

Another issue related with the quality during your calls, is when you can hear beep tones during a call, like if someone is pressing a button on the phone or trying to dial. This is usually known as "talk-off" and the device is interpreting the voice as a DTMF digit.

'''Suggestions to follow:'''

*Upgrade the firmware in your device, sometimes these bugs are fixed in recent versions.

*Change your '''DTMF Tx Method to InBand''' (you have to change this setting in your device and in your account or sub account settings). Test if the DTMF tones are working fine, dial 4747 for this test.

*If Inband doest not work for you, test with DTMF Process INFO and DTMF Process AVT to No, if the options are available in the device.

*Another alternative is as follows: DTMF Tx Method: AVT, DTMF Tx Mode: Strict, DTMF TX Strict Hold Off time: 70.

== Echo during calls ==

'''We have different factors that can cause Echo during the calls, we will review some suggestions to work with:'''

*Check the volume on the phone is not too loud, it is possible the phone is causing the issue.

*Make a call dialing '''4443 for echo test''' and see if you can reproduce the same situation with this test.

*Again, check the '''firmware on the device''', usually this can help to reduce the echo if you do not have the latest firmware.

*The default gain on some devices, is typically too high and can cause echo. For instance on Cisco PAP devices, you can adjust the '''FXS Port Input Gain and FXS Port Output Gain''', one at a time, in increments of three. You can test using -1 and -11.

Note: Input Gain = how you sound to the other party. Output Gain = how the other party sounds to you.

*If the above does not solve your problem, and you have a Linksys device, verify that Echo Canc Enable, Echo Canc Adapt Enable, and Echo Supp Enable are set to Yes. (These are default settings.)

* '''If you use laptop''' (integrated mic/speakers), echo can be caused by microphone catching noise from speakers. Try lowering MIC Input sensitivity.

== One-Way Audio ==

You can hear the other party but they can not hear you, and vice-versa. When a situation like this is present, is know as "one-way audio", and usually is related with the NAT. '''Let's try with the following suggestions:'''

*From the account or sub account settings, select '''always NAT=Yes''' (is the option recommended by VoIP.ms).

*Only as a test, place the device in DMZ, to make sure if the issue is related with the NAT, even if this work, '''do not leave the device in DMZ after finishing troubleshooting.'''

*Is your router is appropriate for VoIP? If you have a router and a modem, try to bypass the router to verify if the issue gets duplicated.

*If your router includes a '''SIP ALG and/or SPI Firewall''' setting please ensure that it is disabled. That setting is common in D-Link and Netgear routers. If this does not help make sure you are using the most recent firmware version for your device.

*If you have an '''ATA device (Linksys)''' you can work on the following settings:

'''Under SIP page.'''
*RTP Packet Size: 0.020
*G729a Codec Name: G729
*G729b Codec Name: G729

'''Under the Line page.'''
*NAT Mapping Enable: Yes
*NAT Keep Alive Enable: Yes
*Preferred Codec: G711u
*Use Pref Codec Only: No

== Contact your provider ==

Could it be the case my quality issue resides on the VoIP provider? Yes, it is possible. Some things we can check and specify to provider when opening the ticket are:

*Are the sound issues present only with '''incoming calls''' or only with the''' outgoing calls''', or both? Are the sound issues present while dialing 4443 to reach echo test?

*If issues are present only when calling certain areas or specific countries.(please provide the number(s) when opening the ticket)

*If the issue are happening only with the incoming calls, then please route your DID to '''echo test''', and call it from an external provider (preferably landline) and check if the issues appear when doing this test.

*If the issue is present with outgoing calls only, make a test using the different options for '''Routing ([http://wiki.voip.ms/article/Value_vs_Premium value or premium])''', if the sound issue happens with one route and not with the other, then you need to contact the provider.

Portions of this article have been taken from "How to Troubleshoot Poor VoIP Audio Quality" by Mango. Used with permission.

[[category:guides]]