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Author Topic: Bookmark showing examples of X_InboundCallRoute & other interesting tips:  (Read 171763 times)
QBZappy
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« on: February 26, 2014, 06:08:30 pm »

I will be using this thread as a bookmark for myself. My memory isn't as good as it used to be, I sometimes have trouble finding old posts with examples. I like easy to understand one liners like these examples from Hortoristic. I have seen many interesting, creative and useful Inbound call route configurations such as the "Oleg Method" used to block spam calling. I will start collecting other examples as I see fit. Anyone can contribute their favorites. Feel free to add to this community "Best of" list. Remember to keep it short with a corresponding short one line explanation of what it does. Perhaps some of these examples could eventually end up in a future version of the admin guide.

Remove any of the hi-lighted text as these were just for comments:

X_InboundCallRoute:
{(16046985518):aa}, -- any call from 16046985518 should get AA
{(13604703652)>(19142961123):sp2(011441903208888)}, -- if 19142961123 was called from 13604703652, then dial out SP2: 011441903208888 - using my voip.ms international plan
{(13604708123)>(19142961111):sp1(13604703562)}, -- if 19142961111 was called from 13604708123, then dial out SP1: 13604703562
{>44113604708123:sp1(13605158499)}, -- if any number calls this UK DID, then dial local cell number: 13605158499 from SP1
{(x|xx|xxx|xxxx|xxxx|xxxxxx|un@@.):}, -- a great SIP scanner avoidance string
{ph}
« Last Edit: March 08, 2014, 10:58:44 am by QBZappy » Logged

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ianobi
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« Reply #1 on: February 28, 2014, 09:34:36 am »

Yes, some good ones there from Hortoristic. I'll throw this one in as it seemed to get some interest from users:

Quote
Call forking with varying delays and using the Oleg Method:
Voice Services > SP1 Service > X_InboundCallRoute:
{>1234567:ph,sp2(12222222222;d=5),sp2(13333333333;d=15)}
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QBZappy
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« Reply #2 on: March 01, 2014, 10:28:10 am »

It should look like this.

DigitMap:
(<S0:9999>|[1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))

OutboundCallRoute:
{(<ob200111111*>9999):pp},{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}

I set this up like you requested, but it might cause you problems. There will be a slight delay before you hear the dial tone from the PBX after the phone is taken off hook. Maybe a second. If you have a phone that you dial a number and then hit SEND, the call will probably fail.

I think a better setup is when you take the phone offhook, you will hear dial tone from OBi110 but any number dialed will be routed to the PBX.

Here is the setup:

OBi110#1

Physical Interfaces -> Line Port -> InboundCallRoute:
{pp(ob200222222)}

Voice services -> OBiTalk Service -> InboudCallRoute:
{200222222>(<#:>):li},{200222222>(XX.):li}

OBi110#2

Physical Interfaces -> Phone Port

DigitMap:
(XX.S4|[1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))

OutboundCallRoute:
{(<ob200111111*>XX.)|(<ob200111111*>#):pp},{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}


Voice services -> OBiTalk Service -> InboudCallRoute:
ph

Dial # to get dial tone on the PBX.
Any other dialed number will be routed to the PBX.
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QBZappy
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« Reply #3 on: March 06, 2014, 12:45:29 pm »

I helped a newbie with a setup and he asked for an explanation of how the InboundCallRoute worked. I thought answering the question in a separate thread might help others.

This is the route he asked about:
{pvpham>(Msp1):sp1},{101>17771234567:aa},{ph}

The above rules are use to make calls from an Android phone using CSipSimple softphone. The first rule uses Sip2Sip service to make single stage calls by bridging an inbound call out SP1. The second uses Callcentric to make calls via the AA.

Rules are processed left to right.
When a rule matches a call, it is immediately routed to the destination.
Each rule has 2 tests and the format is:
{test1>test2:dest}

You can optionally have only 1 test like this:
{test1:dest}  or  {>test2:dest}

If you have 2 tests, both tests must be true to get a match on a rule.
If a test is enclosed in parentheses it is a DigitMap.
If it is not enclosed it is a LITERAL.

Rule {123:dest} is testing for literal "123"
Rule {(1xx):dest} is testing for literal "1" followed by 2 digits
Rule {1xx:dest} is testing for literal "1" followed by 2 alpha "x"

If a test is in this format (Maaa) it uses the DigitMap of a trunk.
Rule {(Msp1):dest} uses SP1 DigitMap for the test.

The tests have different meanings depending if it's a normal inbound call or an inbound call initiated by a softphone/ATA meant to be bridged.

Normal Inbound Call:
Test1 is the CallerID of the inbound call.
Test2 is your UserID of the service provider.

{101>17771234567:aa}

Callcentric sends the extension number as CallerID when a sub-account calls the main account.
101 is the extension number.
17771234567 is your Callcentric UserID.
If both these match the call is routed to the Auto Attendant.

Inbound Call meant to be bridged:
Test1 is your UserID
Test2 is the outbound phone number.

{pvpham>(Msp1):sp1}

pvpham is your Sip2Sip UserID.
(Msp1) is using SP1 DigitMap to match on the dialed number.
If both tests are true the call is bridged out SP1.

If both rules fail to match, the call is routed to the Phone Port.

« Last Edit: March 07, 2014, 06:24:58 am by QBZappy » Logged

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QBZappy
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« Reply #4 on: March 06, 2014, 12:50:23 pm »

Here is a trick you can do to determine the username.
Temporarily add {sp1($2)} to the beginning of your SP2 X_InboundCallRoute, then call your SP2 phone number.
$2 is a variable that contains the username of SP2.
The above rule will attempt to bridge the inbound call out SP1 using the username as the outbound phone number.
This call will obviously fail, but the call history will show the username as the outbound number.
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QBZappy
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« Reply #5 on: March 06, 2014, 12:53:22 pm »

After playing with something ianobi posted I realized that the Obi can allow us to make a skype account simultaneously ring on incoming calls.   In my example we assume sp2 is set up for sip and sp1 is your inbound provider you want to simulring Skype.

Voice Services > SP1 Service > X_InboundCallRoute:
{ph,sp2(skypUserName@skype.jetnumbers.com)}

I can't say how reliable this trick will be.  It needs more testing.  So far so good.  When I call my GV number all forwarding phones ring and my Skype account rings.  If I don't pick up GV Voicemail kicks in.

Here is a little more by ianobi that may be useful in combination with this.

It is possible to make this imitation call hunting a little more sophisticated. Consider an InboundCallRoute such as:

Voice Services > SP1 Service > X_InboundCallRoute:
{ph,sp2(12222222222;d=5),sp2(13333333333;d=15)}

A call coming in to sp1 will ring the phone connected to the OBi. Five seconds later the call will be forked to 12222222222. Fifteen seconds later the call will be forked to 13333333333. All three endpoints will now be ringing, the first to answer takes the call. If a second call comes in while the first call is in progress, then the two free endpoints will receive ringing (after any applicable delays) and either may take the second call.

To use the Oleg Method with this example:

Voice Services > SP1 Service > X_InboundCallRoute:
{>1234567:ph,sp2(12222222222;d=5),sp2(13333333333;d=15)}

Where 1234567 = Voice Services > SP1 Service > SIP Credentials > AuthUserName

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QBZappy
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« Reply #6 on: March 06, 2014, 12:56:39 pm »

It’s obvious from this thread and the one that you quoted originally that your manual is much needed!

Looks like we are making some progress. As you may know, a SIP call comes in two parts: Firstly, the SIP signalling which routes the call, sets it up , disconnects it etc. This we have now achieved. Secondly, the media part (our voices) which rely on Real Time Protocol (RTP) after SIP has set the call up. This seems to be our problem now. There are two ways to improve RTP connection:

1. Ensure that the correct ports are forwarded in your router. The Obihai FAQ section is out of date on this issue. As you are using sp3 as a “carrier” for Voice Gateway 3, in your router you need to forward the range of RTP ports defined here:

Service Providers > ITSP Profile C > RTP > LocalPortMin
to
Service Providers > ITSP Profile C > RTP > LocalPortMax

The protocol defined in your router need only be UDP.

Also if your router has a SIP ALG (Application Layer Gateway) function, then turn it off.

2. Use a STUN server. You can use any public STUN server, I use ideasip as it seems reliable. Set up as follows:

Service Providers > ITSP Profile C > General > STUNServer: stun.ideasip.com
Service Providers > ITSP Profile C > General > STUNServerPort: 3478 (default)

No need to check STUNEnable. I know this sounds strange, but it is not needed and may confuse later providers using sp3 or other voice gateways that use sp3.

One more change:

Voice Services -> Gateways and Trunk Groups -> Voice Gateway3
Name : Voxbeam
AccessNumber : sp3(sbc.voxbeam.com;op=s)
DigitMap : (ppppppp(91xxxxxxxxxx|xx.S4)|<ppppppp>(91xxxxxxxxxx|xx.S4))
AuthUserID : +1xxxxxxxxxx

When the AccessNumber is used op=s tells sp3 (which you are using as a carrier for Voice Gateway 3) to use the STUN server set up on sp3 (ITSP Profile C).

If you do write the manual, then remember that the DigitMap will depend on what numbers the user wants to use Voxbeam to dial. Good luck!


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QBZappy
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« Reply #7 on: March 06, 2014, 12:59:42 pm »

1. This seems odd. When you pick up the phone it's the OBi110 dial tone that you hear. Even if you make PSTN your Primary Line, you will not hear PSTN dial tone by simply picking up the phone. Can you try a different phone? A simple corded phone would be best for testing.

2. A second dial tone can be provided in the Phone Port DigitMap. There is a choice of three dial tones:

di = Dial Tone
di2 = Second Dial Tone
od = Outside Dial Tone

If we use second dial tone the format is like this:

Physical Interfaces > PHONE Port > DigitMap:
([1-9]x?*(Mpli)|[1-9]S9|[1-9][0-9]S9|911|**0|***|#|**1{t=di2}(Msp1)|**2{t=di2}(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))

User picks up phone gets dial tone, then dials **1 and gets the second dial tone. Second dial tone disappears when any further digit is dialled.


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QBZappy
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« Reply #8 on: March 06, 2014, 01:08:10 pm »

It is possible to make this imitation call hunting a little more sophisticated. Consider an InboundCallRoute such as:

Voice Services > SP1 Service > X_InboundCallRoute:
{ph,sp2(12222222222;d=5),sp2(13333333333;d=15)}

A call coming in to sp1 will ring the phone connected to the OBi. Five seconds later the call will be forked to 12222222222. Fifteen seconds later the call will be forked to 13333333333. All three endpoints will now be ringing, the first to answer takes the call. If a second call comes in while the first call is in progress, then the two free endpoints will receive ringing (after any applicable delays) and either may take the second call.

To use the Oleg Method with this example:

Voice Services > SP1 Service > X_InboundCallRoute:
{>1234567:ph,sp2(12222222222;d=5),sp2(13333333333;d=15)}

Where 1234567 = Voice Services > SP1 Service > SIP Credentials > AuthUserName


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QBZappy
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« Reply #9 on: March 08, 2014, 09:18:04 am »

Re: Trouble with Credit Card terminal but faxing works great
http://www.obitalk.com/forum/index.php?topic=6625.msg41922#msg41922

Modem is sensitive to jitter buffer adjustments during a call. Your OBi can detect FAX tone and will switch off jitter buffer adjustment when FAX is detected (it might even switch to T.38/FAX Relay if the call peer also supports it). That makes your FAX work better.

However there is no Modem signal detection support.
There is a new feature in the latest firmware (3.0.1.4109) that may help (do a f/w upgrade first).
It allows you to disable jitter buffer adjustment for the next outgoing call, by dialing a *code (of your choice). This is what you can do: Add a Star Code (under Star Code Profile) like this:

*01, Modem Call, set($Noji1,200),set($Noec1,1),set($Cdm1,3)

The 3 set() commands tell the obi to do the following for the next outbound call:
1. Disable Jitter Buffer Adjustment, and use a fixed jitter buffer length of 200 ms
2. Disable Echo Canceller
3. Use only G711u and G711a codec

Replace *01 with any code you like, as long as it does not conflict with other codes.
Then dial the target number and see if the modem call works better.
Note that this only controls the OBi; the quality will still be affected by the behavior of the call peer and any network impairment. Good luck.

Tag: How to control Jitter Buffer Adjustment
« Last Edit: April 23, 2014, 05:08:32 pm by QBZappy » Logged

Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
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« Reply #10 on: March 08, 2014, 09:22:58 am »

How to call between Ph1 and ph2 like an intercom

I have an Obi202 with two VOIP lines connected in my house. However, I'd like to be able to call between them (use it as an in-house intercom) w/o using the VOIP service, since the VOIP service charges a per minute fee. Can you do that, and if so, how? I assume you can use the ObiTalk Network to call for free, but since both phone lines come out of the same Obi202 device, I thought maybe there was a simpler way?

Thanks in advance for your help.

Press # on the phone.  How is that for an easier way?  Smiley
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QBZappy
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« Reply #11 on: March 10, 2014, 05:06:19 pm »

Provisioning Utility for OBi100/110/200/202/300/302
http://www.dslreports.com/forum/r28450653-Provisioning-Utility-for-OBi100-110-200-202-300-302

http://www.dslreports.com/forum/r29093546-How-is-Vestalink-able-to-set-up-my-Obi-through-their-web-site-
If you'd like to have backup files that include your Passwords and PINs, you can create them with OBiProv:

1. Use OBiProv to create a provisioning file which contains Passwords and PINs, leaving the 'K' and 'iv' fields blank so that no AES or RC4 encryption occurs.

2. Rename the resulting MACADDR-1.xml file to MACADDR.xml.gz and decompress it using gzip.exe: gzip -d MACADDR.xml

3. The decompressed MACADDR.xml file may then be used to totally restore an OBi, including Passwords and PINs.
« Last Edit: March 10, 2014, 05:09:44 pm by QBZappy » Logged

Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
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« Reply #12 on: March 11, 2014, 07:03:27 am »

Spoofing Solved
http://www.obitalk.com/forum/index.php?topic%3D4658.0
I never could get spoofing to work.  I gave it one more try and I did it!

This works WITHOUT enabling X_SpoofCallerID.  The problem with X_SpoofCallerID is that some providers don't like spoofing and when enabled it's on  for everything going out SPx.

I got it to work with PBX in a Flash and Callcentric.

Here's an InboundCallRoute example:

{ph,sp2(100@192.168.1.999:5060;ui=$1),sp2(17772223333@in.callcentric.com;ui=$1}

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QBZappy
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« Reply #13 on: March 16, 2014, 04:14:17 pm »

http://www.obitalk.com/forum/index.php?topic=4067.msg35778#msg35778

SIP scanners (oleg method) From the man himself.

I think many questions were already answered, will try to clarify some...

Early suggestions (in this thread) to block scanners were based on matching caller. Caller is a string sent by calling party, that's how it represents itself. Provided that normally you receive calls with valid 10 digits caller id and that scanners use something like "10001", "admin", etc. you can filter them out. But what if scanner uses a right pattern? Your OBi will ring the phone...

My idea was to use callee id instead. The way is documented in Inbound Call Route part of OBi manual  (may be lacking more examples). Most of us use one or several providers, may-be receiving direct SIP calls - pretty much determined set of valid strings. All other calls may be dropped. That's what I made.
BTW, old good Sipura / Linksys adapters allow the only user_id, any call not matching it does not ring.

>('myname'|123456@.):ph1
This is to allow incoming calls to myname@myhost (direct SIP calls) and 123456_me@myhost (voip.ms pattern, all sub-accounts have the same prefix), forward both to ph1 (I do not use ph2 now) and disregard all other calls. Note that you may receive SIP calls from several providers on the same SPn, that's why you may want to combine several patterns. You need separate SPn only if you want to register with several providers.

>('1777xxxxxxx'):ph1
This should work for Callcentric, but I believe you may remove quotas and even parentheses.

>>> Aside from the nerdiness what advantage does one gain using SIP URIs vs the traditional way...
It may be independent from any provider (sometimes more reliable), completely free, always the best quality (direct traffic)... Not something necessary though...

>>> 3/20/2013 2:19:10 AM    INVITE sip:+972592280470@12.34.56.78:5078 SIP/2.0
This is from syslog. I have it enabled on OBi, most verbose and sent to the server.
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QBZappy
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« Reply #14 on: March 19, 2014, 09:50:40 am »

USE SKYPE ON YOUR OBI: A HOW-TO GUIDE (Google Voice alternative)
https://www.obitalk.com/forum/index.php?topic=3920.0

INTRODUCTION:  The following is a how-to guide on setting up your OBIHAI phone for in/outgoing calls via Skype.  It assumes no prior experience and is written for users, like me, who are less savvy with telephony.  Virtually all the information contained below was first generated in a post by RonR.  Several users commented on and clarified some aspects of this.  I spent several hours working through the original, more advanced post.  I am writing this to give a step-by-step “idiot’s guide” to be helpful to others.

OVERVIEW:  In a nutshell this guide will enable the user to place and receive calls on their OBIHAI via Skype.  It uses a free third-party software called SipToSis along with the implementation for the OBIHAI provided by RonR.  For your reference here is the original post:  http://www.obitalk.com/forum/index.php?topic=1196.0

PROCEEDURE:
1) Per RonR:
Quote
Go to http://www.mhspot.com/sts/siptosis.html and download SipToSis, Sip to Skype integration software.  Extract the archive to a folder named SipToSis.  Execute SipToSis_win.bat.  When activity has stopped, terminate SipToSis.

2)  In that downloaded folder edit two documents a) SkypeOutDialingRules.props and b) SkypeOutDialingRules.props by opening them in Notepad and at the very bottom of the specified document pasting the lines in bold below:

  a.   SkypeOutDialingRules.props:

    i.   [This is optional.  You are creating up to ninety-nine speed dials for Skype user IDs you may wish to call.  If you normally call land lines and don't need to reach users by their Skype ID you can omit this step.  You can also do it later]

Per RonR:
Quote
Add up to 99 Skype UserID's as Speed Dials in SkypeOutDialingRules.props:

^1$:SkypeUserID1
^2$:SkypeUserID2
^3$:SkypeUserID3
^4$:SkypeUserID4


^96$:SkypeUserID96
^97$:SkypeUserID97
^98$:SkypeUserID98
^99$:SkypeUserID99
 

ii. Per RonB
Quote
Add US/Canada dialing support in SkypeOutDialingRules.props:

^(1[2-9][0-9]{2}[2-9][0-9]{6})$:+$1
^011([0-9]{7,})$:+$1
 

  b.   SkypeToSipAuth.props:

    i.   You will need to configure your OBIHAI for a new “service provider” [the internal instructions within your OBIHAI set up] for use with Skype.  For this you need to add your OBIHAI's IP address to that configuration.  OBIHAI service providers are labeled SP1-SP4.  Likely, your SP1 is already set up for Google voice in which case you would use SP2, SP3 or SP4 for your Skype calls.  This guide will assume SP2.   [Note what you will be doing is setting up SP2 as SIP, however you do not need an actual sip account because siptosis replaces that in this instance].  Using notepad open the SkypeOutDialingRules.props file and then paste the following entry to the bottom after the last line of text:

*,sip:siptosis@192.168.1.150:5061,YourSkypeUserID     

NOTE: replace “192.168.1.150” with your OBIHAI's IP address which you can find by dialing *** and then "1" from the phone connected to the Obi.  Use either “:5060” if you are setting up SP1, or “:5061” if you are setting up SP2.  Normally you will use the latter.  If you're setting up SP3 or SP4 you'll have to research which port number to use.  Replace “YourSkypeUserID” with your actual Skype username.

    ii.   There is an error in the provided file: at the bottom of the text there are two lines which read:
#Default: all incoming skype callers get the invalid destination message
*,play:clips/invalidDest.wav
    There should be a #added in front of the second line so it now reads:
#*,play:clips/invalidDest.wav
This error was pointed out by user Veronica, many thanks!


3) Everything from this point on you'll do to the OBIHAI.  There are two ways of doing this, one is to browse to your OBIHAI's IP address from your local computer, the other is to log onto OBItalk on the Internet and configure through that.  RonR prefers the local method but other posts including from OBIHAI tech support suggest everything should be done through OBItalk.  The following post explains this in more detail, but the gist of it is that anything you do in OBItalk overwrites custom settings done locally.  Therefore, some of the following description will differ slightly from what RonR wrote because it uses the OBItalk configurator.  See this post if you want to understand more about these differences:

http://www.obitalk.com/forum/index.php?topic=61.msg109#msg109

  a)   As stated in the article there is one thing you can only do locally, which is to backup your existing configuration files.  This should be done first as follows:
   
     i.   type your OBIHAI's IP address into your web browser in the same network where the OBIHAI resides.  It should take you to a login page requesting username and password.  Note: depending how your OBIHAI is set up (technically speaking whether you're coming in from the WAN or LAN side of its dual network plugs connections you may need to modify the default setting which prevents WAN access [even though we are actually coming into it locally, for some reason OBIHAI treats this as WAN access].  To enable WAN access on your OBIHAI phone press the following keys: 

* * * 0, then 30, then 1 if it is not enabled (to change it to being enabled)

    ii.  Once you access the OBIHAI locally, type the default username and password which is admin/admin [unless you have changed it].
   
    iii. Click system management on the left and then device update below that.  Then press the appropriate buttons to backup your user prompts and your backup configuration.  Under backup configuration I'm not sure which boxes to check.  If you're worried about that make more than one backup with different boxes checked.  You could also read more in the administration guide on this page if desired: http://www.obihai.com/docs-downloads.html

    iv. Save those backup files in case you need to restore.  Note, if your OBIHAI is in default configuration you probably don't need to do this backup because on the same tab as where you backup you'll notice there's a button which will restore all default settings.  However, it good to know about this backup procedure because as you modify the OBIHAI you will likely want to make some backups along the way.

  b)   Everything else will now be done in OBitalk.  Exit the local browser and Login to OBItalk and proceed as follows:

    i.   Push blue button for OBIHAI expert configuration.  Click yes at the warning and click enter OBIHAI expert on the next page.  You will note the expert configuration is almost identical to the local configuration page where you backed up the user settings.  However, as stated above if you make changes in both locations the ones done in OBItalk will overwrite the ones done locally.  That is why we're configuring through OBItalk.

    ii.   Add a Voice Gateway for dialing: Go to Physical Interfaces -> PHONE Port [Phone 1, 2, or both depending which you want to configure for Skype.  I did it for phone 1] -> DigitMap : ...

Here you'll modify the entry.  You can either add the bold/underlined portion of the following notation to the end of your existing DigitMap text:

“...|#1(Mvg8)|(Mpli))”,

Or, you can paste this entire entry replacing what you have now:

([1-9]x?*(Mpli)|[1-9]|[1-9][0-9]|911|**0|***|#|**1(Msp1)|**2(Msp2)|**3(Msp3)|**8(Mli)|**9(Mpp)|#1(Mvg8)|(Mpli))

    iii.   Go to:  Physical Interfaces -> PHONE Port [Phone 1, 2, or both depending which you want to configure for Skype.  I did it for phone 1] -> OutboundCallRoute : ...

Here you'll modify the entry.  You can either add the bold/underlined portion of the following notation to the end of your existing OutboundCallRoute:

… ...,{(<#1:>(Mvg8)):vg8},{(Mpli):pli}

Or, you can paste this entire entry replacing what you have now:

{([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**3:>(Msp3)):sp3},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(<#1:>(Mvg8)):vg8},{(Mpli):pli}

    iv.   Go to Voice Services -> Gateways and Trunk Groups -> Voice Gateway8 and fill-in the lines as designated below:

Name : SipToSis

AccessNumber : SPx(Your-PC’s-IP-address:5070)     // where x= the service provider # 1-4 you are configuring, and where  “Your-PC's-IP address” is the IP for local computer that will be running Skype, note: “: 5070” indicates the port number which siptosis will use.  5070 is the default.  Use it in less you know how and why to change it.

In the DigitMap field, copy/paste: (x|xx|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>[2-9]xxxxxx|011xx.)
where “aaa” is your local area code

AuthUserID : skypests   [note: use this actual user ID]

4.   On your computer open Skype and login to the same account as you provided above.  On that same computer go back to the siptosis downloaded folder and execute (double-click) the SipToSis_win.bat.  file.  Leave the siptosis DOS window open.

Incoming Skype calls should ring the OBi's PHONE Port in addition to Skype on your local computer.

To make a Skype outgoing call on your OBIHAI phone, dial #1 <Skype Speed Dial> # or #1 <Phone Number> #.  [Note: I'm not sure you need the final #after the number but this is how Ron wrote it so I'm repeating it that way]

If you did everything per this plan (and I made no errors copying and expanding upon Ron's instructions) everything should work.

5.   Repeat steps 3A, above to again backup your new OBIHAI configuration.  Remember you have to do this through the local browser access, not through OBItalk.  You may also want to backup the two files you edited from the siptosis downloaded folder.

Good luck, have fun and tell me if I screwed up.
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
Hero Member & Beta Tester
*****
Posts: 2316



« Reply #15 on: March 24, 2014, 05:41:12 am »

How to find info/owner of DID number:
http://www.infodids.com/

How to find info/owner of toll-free DID number:
http://www.800forall.com/
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
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*****
Posts: 2316



« Reply #16 on: March 26, 2014, 04:22:55 pm »

Interesting thread discussing Rollover lines:
http://www.obitalk.com/forum/index.php?topic=1533.0

I couldn't find any info on this searching the forum.

Is there a way to do rollover lines with the obi110?  In other words, if you have 4 POTS lines connected to a multiline phone, and you wanted to port those 4 lines to 4 obi110 units (google voice?), can you make it so that if someone calls your main phone number, it bumps to your next number, and then your next number, and then your next number (or in other words, bumps to your next obi, then the next, then the next).

Or perhaps there is a different way to deal with entirely.
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
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Posts: 2316



« Reply #17 on: March 26, 2014, 04:30:05 pm »

HOWTO: Thwarting SIP Scanners during Set-up
http://www.obitalk.com/forum/index.php?topic=5467.0

Note: Things changed for the better about June/July 2013. OBiTalk has been implementing method 4, Oleg method described below, for at least some of the SIP providers by default. If your provider is not one that OBiTalk lists or if you get a SIP scan, or if you have overridden the X_InboundCallRoute so that OBiTalk does not control the field, or if you choose to not use OBiTalk, the information below will still apply. (note #j)
      =========The need for the following has been reduced========

An unfortunate thing we all must learn about is how to thwart SIP scanners.  It is best to handle this now rather than getting motivated at 4AM by a phantom call with caller ID "1001" or some such.  There are four ways to address the SIP scan intruders.  For those using OBiTalk to set things up (as I do), each involves using the "OBI Expert Configuration" to change values from the defaults.  Methods (pick one only for each SPn Service that is not set up for GV):

0. Disable any unused SPn Service by un-ticking (Voice Services)SPn Service->Enable. This may be unneeded, but it will not hurt for an unused SPn

1. Filter on the length of the caller ID string. This is more complex and seems less effective IMHO.  It would seem easy to circumvent by spoofing a real number.

2. Change the (Voice Services)SPn Service->X_UserAgentPort ports for each SPn  to a number not in the 506x range. This seems effective based on people's experience, but how long until the scanners broaden their scans?

3.  Change the X_AccessList value for each service to permit only the listed IP addresses to be used as a SIP server.   If using ObiTalk expert configuration, that path would be :
 System Management -> Service Providers-> ISTP Profile n SIP -> X_AccessList

 The list is limited in the OBi boxes to 512 characters, and each item in the list can be up to 16 characters long with an average of 14.28 while we are using IPV4 addressing.  This would permit maybe 36 addresses to be listed in a comma-separated list.  This will be plenty for some SIP suppliers, but not enough for others.  I presume Anveo uses anycast IP addresses, because they seem to only use one IP address for each server location.  So with Anveo, set the corresponding X_AccessList to
72.9.149.69,67.212.84.21,176.9.39.206
if you want to allow the US, Canadian and German servers. I tested that one. For voip.ms, the expanded 256-character string would be
Code:
174.34.146.162,173.208.83.50,174.137.63.206,174.142.75.171,174.34.146.162,198.144.158.125,199.21.149.36,209.62.1.2,5.77.36.136,64.120.22.242,67.205.74.164,67.205.74.184,67.205.74.187,67.215.241.250,68.233.226.97,69.147.236.82,74.54.54.178,74.63.41.218,78.129.153.20


This was my preferred method if the IPs are known and fit the string. However if OBiTalk performs method 4 for you automatically, then I would now go that way. I would check to see what OBiTalk did for each SPn before implementing new methods:

4. Method 4 , as discovered by Oleg and applied by Mango, is to change the (Voice Services)SPn Service->X_InboundCallRoute: from {ph}, if that is what it currently is, to
 {>('Insert your AuthUserName here'):ph}
This method is particularly useful for those using SIP providers that have too many IP addresses to use method #3, although it may become the preferred method for more providers. The AuthUserName can be read from your OBi or ObiTalk expert at
(Voice Services)SPx Service->AuthUserName.

Legend for examples:
6065556789 Sample trusted phone number
17771234567 Sample account name/number for SIP provider. I think they are saying for Callcentric they all start with 1777 followed by 7 more digits. Anveo is also all numbers, but
is a little shorter.

Here are some examples of the Oleg method (please suggest more examples and point out needed fixes):

OBi110 with no trusted caller set up with service 1 being routed to the phone port; (Voice Services)SP1 Service->X_InboundCallRoute:

{ph}                 [before]
{>17771234567:ph}  [after adding protection from SIP scanners]


OBi202  with one trusted caller with service 2 being routed to phone line 2; (Voice Services)SP2 Service->X_InboundCallRoute:
{(x.6065556789):aa},{ph2}  [before]

These are both tested to work:
{(x.6065556789):aa},{>17771234567:ph2}  [ Tested and works]
{(x.6065556789):aa},{>('17771234567'):ph2}  [Permits alphanumerics]
[The single quotes are not needed if the ID is all numbers, but will not hurt]

See note #h below.



Notes:

 a. http://www.obitalk.com/forum/index.php?topic=4067.0 discusses the methods including Callcentric IP numbers. http://www.obitalk.com/forum/index.php?topic=5455.msg35330 also discusses the methods including Anveo IP numbers.

b.http://www.obitalk.com/forum/index.php?topic=3544.0 is a feature request asking that OBi firmware to allow specifying blocks of IP numbers into the X_AccessList lists.

c.  http://www.obitalk.com/forum/index.php?topic=4873.0 cites #b plus it asks that the OBi have an option to "Reject SIP requests except from registration server".

d. The motivation for the SIP scanners appears to mostly be probing for PBX servers to allow expensive international and premium calls for profit. http://en.wikipedia.org/wiki/Premium-rate_telephone_number

e. I do not know if a port set up for  Google Voice could get hit with the phantom phone calls, since it uses a different protocol. Changing  X_UserAgentPort will not hurt, and it may be proactive against somebody using that scanning with that protocol in the future.

f.  Unused ports should be disabled. One way is to un-check the (Voice Services)SPn Service->Enable box, which is Enabled by default.  The OBi202 has 4 SPn ports.  Obi100 and OBi110 have 2 SPn ports.  You could change your X_AccessList to 0.0.0.0 as a sure method.

g. Search for "SIP scanners" here or elsewhere for more information.

h. Method 4 was posted by Oleg, and Mango recognized the general applicability of this method. See http://www.obitalk.com/forum/index.php?topic=4067.80 to follow the birth and development of this method. I hope to add more examples for other SIP providers, but it is the Callcentric users who cannot use method #3. I got a hub to use Wireshark packet sniffer to monitor communications to my OBi202. Formats are similar to what Oleg saw for Callcentric. Normally you would only use one method for each SP.

j. The implementation of anti-scanner method by OBiTalk was first posted by ianobi on reply 7 of http://www.obitalk.com/forum/index.php?topic=6266.0

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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
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*****
Posts: 2316



« Reply #18 on: April 11, 2014, 02:33:47 pm »

Voip bandwith calculator
http://www.bandcalc.com/
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
QBZappy
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*****
Posts: 2316



« Reply #19 on: May 15, 2014, 12:27:34 pm »

Home and Remote Configuration Using OBi202s
http://www.obitalk.com/forum/index.php?topic=8003.0

Here’s another version of a master/slave setup which provided a working solution for an OBi owner. The Home OBi202 has two voip numbers; the first dedicated for incoming and outgoing calls on sp1 used by Phone Port 1 only, the second dedicated for incoming and outgoing calls on sp2 used by Phone Port 2 only.

The problem is to replicate the exact same setup at a remote site seventy miles away. The voip provider does not allow multiple registrations on each account.


Remote OBi202                                               Home OBi202

                                                                                   Ph1
                                                                                     |
Ph1 – sp1 ----                                                ---------- sp1 ------ voip 1
                      |------OBiTALK network---------|
Ph2 – sp2 ----                                                ---------- sp2 ------ voip 2
                                                                                     |
                                                                                   Ph2


Home OBi202 – 500111111
Remote OBi202 – 500222222


Home OBi202

Voice Services > SP1 Service > X_InboundCallRoute:
{ph,pp(ob500222222***1)

Voice Services > SP2 Service > X_InboundCallRoute:
{ph2,pp(ob500222222***2)}

Voice Services > OBiTALK Service > InboundCallRoute:
{(500222222)>(<**1:>(xx.)):sp1},{(500222222)>(<**2:>(xx.)):sp2},{500222222:aa},{ph}


Remote OBi202

Set up “fake” SIP providers on sp1 and sp2 (see below). Configure Phone 1 for outgoing and incoming on sp1 only. Configure Phone 2 for outgoing and incoming on sp2 only.

Physical Interfaces > PHONE Port 1 > OutboundCallRoute:
Find this rule {(Mpli):pli} – it’s normally the last rule – delete it and replace with {(<ob500111111***1>(Mpli)):pp}

Physical Interfaces > PHONE Port 2 > OutboundCallRoute:
Find this rule {(Mpli):pli} – it’s normally the last rule – delete it and replace with {(<ob500111111***2>(Mpli)):pp}

Voice Services > OBiTALK Service > InboundCallRoute:
{500111111>(**1):ph},{500111111>(**2):ph2},{ph}


Notes

1. For clarity I have omitted any references to the OBiON 290xxxxxx number. Also I have not included the Oleg Method.
2. The final rule in Voice Services > OBiTALK Service > InboundCallRoute can be as shown {ph} or {ph2} or {ph,ph2} or omitted, depending on how you wish to route general calls on the OBiTALK network to be routed in each OBi202.
3. By setting up suitable speed dials, any Home or Remote OBi Phone Port can call any other Home or Remote OBi Phone Port. This also allows for call transfer between any of the four Phone Ports and three-way calls to be set up.
4. This setup effectively gives each voip line two endpoints. Most voip providers allow two sessions (calls) on each line. This allows a call from the Remote OBi202 from Phone Port 1 to take place at the same times as an ongoing call taking place from Phone Port 1 of the Home OBi. The possible downside of this is that if a call is ongoing at either site an incoming call on the same line will automatically be routed to the other OBi202. This may be good or bad depending on what you want this setup to do. This behaviour can be controlled by changing the “MaxSessions” setting.
5. The Remote OBi202 can be anywhere in the world. For this reason I have not suggested how to provide 911 service to it. If required, then 911 calls can be routed from the Remote OBi202 through the Home OBi202.

Example “Fake” SIP provider for sp1:

Service Providers -> ITSP Profile A -> SIP -> ProxyServer : 127.0.0.1

Voice Services -> SP1 Service -> Enable : (checked)
Voice Services -> SP1 Service -> AuthUserName : 1234 (any numbers/letters)
Voice Services -> SP1 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP1 Service -> X_ServProvProfile : A
Voice Services -> SP1 Service -> CallerIDName : Whatever

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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
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