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Using SipToSis to Make and Receive Skype Calls Through an OBi

Started by RonR, July 19, 2011, 04:08:59 PM

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RonR

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.


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

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


Add the OBi's SPx IP address in SkypeToSipAuth.props (SPx must be configured as SIP):

*,sip:siptosis@192.168.1.150:5061,YourSkypeUserID     // :5060 = SP1 | :5061 = SP2


Add a Voice Gateway for dialing:


Physical Interfaces -> PHONE Port -> DigitMap : ...|#1(Mvg8)|(Mpli))

Physical Interfaces -> PHONE Port -> OutboundCallRoute : ...,{(<#1:>(Mvg8)):vg8},{(Mpli):pli}


Voice Services -> Gateways and Trunk Groups -> Voice Gateway8:

Name : SipToSis
AccessNumber : SPx(192.168.1.100:5070)     // PC's IP address (SPx must be configured as SIP)
DigitMap : (x|xx|1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|<1aaa>[2-9]xxxxxx|011xx.)
AuthUserID : skypests

where aaa is your local area code


Execute SipToSis_win.bat.

Incoming Skype calls should ring the OBi's PHONE Port in addition to Skype

To make a Skype outgoing call, dial #1 <Skype Speed Dial> # or #1 <Phone Number> #.

VulcanTourist@gmail.com

This looks difficult to configure.  It requires the advanced mode only?  I would be very hesitant to try this unless I knew exactly what I was doing.

QBZappy

VulcanTourist@gmail.com,

RonR's setup  is a basic configuration. Between these instructions and the SIPtoSIS website it is not too difficult to setup. However if you want to get fancy, the SIPtoSIS website offers a lot of configuration options which can then get a little confusing. I think RonR did not add more explanations in his post in order to avoid overwhelming novice users.

Maybe RonR can supplement this setup guide with a new paragraph explaining what the DigitMap and OutboundCallRoute is handling as background info so that everything related to OBi+SIPtoSIS is in one posting.
Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.

RonR

Quote from: VulcanTourist@gmail.com on July 25, 2011, 07:19:41 AM
This looks difficult to configure.  It requires the advanced mode only?  I would be very hesitant to try this unless I knew exactly what I was doing.


The changes to SipToSis are quite simple (editing two files: SkypeOutDialingRules.props and SkypeToSipAuth.props to add a couple of lines).

The changes to the OBi are also minimal and are to places that are probably set to Default at the moment.  If you run into difficulty, reversing the changes is simply a matter of replacing the checkmark in the Default checkbox.

JoeinForum

Has anyone tried this and got it to work. I mean by just following the instructions above. I am a newbie and am just trying to follow the instructions but really not getting very far.

For instance, "SPx must be configured as SIP"... how does one go about doing this? Do I need a Skype Sip account?

If someone knowledgeable can expand on these instructions it would be highly appreciated.


RonR

Quote from: JoeinForum on October 01, 2011, 09:41:23 AM
Has anyone tried this and got it to work. I mean by just following the instructions above. I am a newbie and am just trying to follow the instructions but really not getting very far.

I use it many times a day, so I'm confident it works reliably.

Quote from: JoeinForum on October 01, 2011, 09:41:23 AM
For instance, "SPx must be configured as SIP"... how does one go about doing this? Do I need a Skype Sip account?

Assuming SP2/ITSP Profile B is not in use, it can be configured for SIP without having a SIP provider/account:

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

Voice Services -> SP2 Service -> AuthUserName : (put anything here)

Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)

Voice Services -> SP2 Service -> X_ServProvProfile : B

Quote from: JoeinForum on October 01, 2011, 09:41:23 AM
If someone knowledgeable can expand on these instructions it would be highly appreciated.

If you have additional questions, feel free to ask.

Dav3yDark0

Quote from: JoeinForum on October 01, 2011, 09:41:23 AM
Has anyone tried this and got it to work. I mean by just following the instructions above. I am a newbie and am just trying to follow the instructions but really not getting very far.

For instance, "SPx must be configured as SIP"... how does one go about doing this? Do I need a Skype Sip account?

If someone knowledgeable can expand on these instructions it would be highly appreciated.



I do not have this setup myself as I have a standard phone with Skype support that I use. But what RonR was referring to is that in the OBi device's configuration, SP1 or SP2 (Service Provider 1 or 2) needs to be configured for SIP in the Signaling Protocol, rather than Google Voice. At least 1 SIP provider needs to be configured for SIP to work on Voice Gateways, if both providers are configured for GV it will not work.

Dave

RonR

Quote from: Dav3yDark0 on October 01, 2011, 10:15:07 AM
At least 1 SIP provider needs to be configured for SIP to work on Voice Gateways, if both providers are configured for GV it will not work.

Just to clarify...

You do not need a SIP provider configured on the OBi for SIP to work on Voice Gateways.  At least 1 Voice Service (SP1/SP2) needs to be configured for SIP protocol, but this can be accomplished without a SIP provider as noted in Reply #5 above.

JoeinForum

Ok wow! thanks for all the responses! I am encouraged to try some more. RonR thanks for your offer!!

So right now I am at the point where if I make a call to my skype number I get a message saying invalid destination. The siptosis command window says call received, in progress, play invaliddest.wav and then call finished.

I was not sure about the Phone Port - Digit Map part... I copied the working entries from this post by RonR  http://www.obitalk.com/forum/index.php?topic=701.msg0;topicseen#new

Outgoing calls... return 40 error.


RonR

Quote from: JoeinForum on October 01, 2011, 12:42:29 PM
So right now I am at the point where if I make a call to my skype number I get a message saying invalid destination. The siptosis command window says call received, in progress, play invaliddest.wav and then call finished.

Did you configure SipToSis?:

*,sip:siptosis@192.168.1.150:5061,YourSkypeUserID

using your OBi's LAN IP address in place of 192.168.1.150 and the appropriate SIP port (:5060 = SP1, :5061 = SP2)

Quote from: JoeinForum on October 01, 2011, 12:42:29 PM
I was not sure about the Phone Port - Digit Map part... I copied the working entries from this post by RonR  http://www.obitalk.com/forum/index.php?topic=701.msg0;topicseen#new

Outgoing calls... return 40 error.

All you really needed was the addition from the original post of this thread.

Did you add one or more Skype Speed Dials in SkypeOutDialingRules.props?:

^1$:SkypeUserID1

replacing 'SkypeUserID1' with the Skype UserID you wish to have called when Skype Speed Dial 1 is called (#11#).

JoeinForum

Thanks again RonR for your help.

Yes, I had done those two things and have now inserted the bold text into the phone port - digimap as your the original post indicated. Now I get further with calling out. When I dial from my handset... skype opens on my computer and actually dials the number I call. But then nothing happens and it finally disconnects. In coming calls have the same problem as before

I will detail what I have done so far- possibly something obvious missing

Download and extract SiptoSis - modified two files
Add Skype UserID's as Speed Dials in SkypeOutDialingRules.props:
Add the OBi's SP2 IP address (obtained from my router DHCP) in SkypeToSipAuth.props
*,sip:siptosis@192.168.0.105:5061,MYSkypeUserID



Went to OBI dashboard selected second item
SP1 is google voice: Here it says "connected"
SP2 edited with the following: Here it says "Register Failed: 501 Not Implemented (server=127.0.0.1:5061; retrying)"

SP proxy server: 127.0.0.1
SP proxy server port: 5061  
Outbound Proxy Server: blank
Outbound Proxy Server Port: 5061
User Name: skypename
Password:skype passwd    
URI : blank

Then went into OBI Expert and

1. Physical Interfaces -> PHONE Port -> DigitMap : ...|#1(Mvg8)|(Mpli))
Physical Interfaces -> PHONE Port -> OutboundCallRoute : ...,{(<#1:>(Mvg8)):vg8},{(Mpli):pli}

2. Voice Services -> Gateways and Trunk Groups -> Voice Gateway8:
I've literally written whats in the quotes
Name : "SipToSis"
AccessNumber :"SP2(192.168.0.104:5070)"  104 is my laptop
DigitMap : "(xx.)"
AuthUserID : "skypests"

3. ITSP Profile B SIP did the following as per yourpost

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
In this section:
Here ProxyserverPort and OutBoundProxyPort are 5061
RegistrarServerPort is 5060

4. Under Coiver service sp2
Voice Services -> SP2 Service -> AuthUserName : skypename
Voice Services -> SP2 Service -> X_ServProvProfile : B


RonR

I've got to run out for a while, but let me throw a couple of things out...


SP2/ITSP Profile B should have ALL settings at Default except for:

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

Voice Services -> SP2 Service -> AuthUserName : (put anything here)

Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)

Voice Services -> SP2 Service -> X_ServProvProfile : B

SP2 status should show : Registration Not Required

Physical Interfaces -> PHONE Port -> DigitMap:

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

Physical Interfaces -> PHONE Port -> OutboundCallRoute:

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

JoeinForum

Thanks again RonR, I'm posting my latest situation here and will wait till you get back. Thanks again for all your help in trying to get this sorted out. Its highly appreciated.

To answer your last post ...Yes, I have all that as you have instructed.

Except this...
SP2 status should show : Registration Not Required



When I first enter the Device Configuration page I have two entries under Voice Services

Sp1 is google
and sp2 is what I am working on. I am not sure what the settings should be for that... when I click edit and am taken to a new menu

SP proxy server: 127.0.0.1
Service Provider Proxy Server Port: should this be 5060 or 5061
Outbound Proxy Server: blank
Outbound Proxy Server Port:should this be 5060 or 5061
User Name: skypename
Password:skype passwd   
URI : blank

These seem to populate the SP2/ITSP Profile B at least as far as choosing 5060 or 5061

RonR

You need to be making all these changes from the OBi Expert area if you're using the OBiTALK Web Portal.  I don't use the OBiTALK Web Portal as I feel it's more work than simply configuring the OBi directly, so you're on your own with it.  If you should decide to configure the OBi directly, you must disable Auto Provisioning:

System Management -> Auto Provisioning -> ITSP Auto Provisioning -> Method : Disabled

Whichever route you take, all Service Providers -> ITSP Profile B and Voice Services -> SP2 Service settings should have the Default box checked except for the ones I noted.

JoeinForum

I see... I was starting to wonder if that was where you were configuring it from. Yes, I was in the expert area.

But now, I have done this
System Management -> Auto Provisioning -> ITSP Auto Provisioning -> Method : Disabled

and will attempt to configure directly as per your suggestion.

Need to take a break and come at this fresh. Will post more tomorrow. Thank you again for all your assistance. Hopefully, if I can get this to work will make a detailed set of instructions suitable for other newbies.

ccclapp

hi all,

Thanks for this thread.  I've been trying to follow these instructions and has made some progress but not enough to get to the finish line.  Here are my specific questions/problems:

QuoteAdd the OBi's SPx IP address in SkypeToSipAuth.props (SPx must be configured as SIP):

*,sip:siptosis@192.168.1.150:5061,YourSkypeUserID     // :5060 = SP1 | :5061 = SP2

which IP address to I use, the one listed under LAN or WAN?  For your reference I only have the Internet port on my OB connected.  Thus, the only meaningful IP I have is the WAN IP.  This is the same as what my Comcast router shows as it's given DHCP address for the OB.having said that, in the OB web access under router configuration >  LAN settings I see a different IP address.  I have tried both in the SkypeToSipAuth file.  Please note: when I run the BAT file everything seems fine except my second to last line always says "STUN: error – unknown host: stun.xten.net


Next, I make these edits:
QuoteSP2/ITSP Profile B should have ALL settings at Default except for:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Voice Services -> SP2 Service -> AuthUserName : (put anything here)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B
SP2 status should show : Registration Not Required

I boot after every edit.  Following this for SP2 service status initially I get "registration not required", but after a minute or two the status changes to "service not configured".  Ivory tried this several times and it always reverts back to not configured status.

I have also copied and pasted your full digimap lines but continue to have the two problems listed above.

When you say
QuoteVoice Services -> SP2 Service -> AuthUserName : (put anything here)
do you literally mean put anything there or does it have to be something meaningful?

I'm trying to literally follow your directions Word for Word as this is somewhat foreign to me.  Assuming you've truly listed all the steps and I can copy and paste your digital map text in full, I'm not sure what I'm doing wrong.

Have you omitted any steps or assumed the user has some independent knowledge, because I am just copying exactly what you wrote.

Thanks very much it sure would be great to get this working!

QBZappy

Quote from: ccclapp on August 15, 2012, 09:42:28 PM
Please note: when I run the BAT file everything seems fine except my second to last line always says "STUN: error – unknown host: stun.xten.net

Go to command prompt and ping stun.xten.net (times out)
Now try this stun server: ping stun.sipgate.net (this one returns TTL info)

List of stun servers: http://www.voip-info.org/wiki/view/STUN
Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.

ccclapp

Hi

Thanks for the reply.

Im sorry, but I dont understand what you are saying and why.  I am very new at all of this.  Isnt my issue likely what IP I should be using in the SkypeToSipAuth.props file edits?

I am trying to use SipToSis for Skype Calls Through an OBi.  I had questions about the directions posted. What would pinging a sipgate address have to do with this? 

Sorry, I dont understand.

Thanks


QBZappy

ccclapp,

You are right, your main issue is something else.
Just pointing out that the stun server stun.xten.net does not seem to be working. The ping was just how to test if the stun server would reply.

I haven't used siptosis in over a year. Try entering the ip address returned by dialing ***1 on the phone connected to the OBi.
Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.

ccclapp

GOOD NEWS, IT'S WORKING (99%)!!  Lots of thanks to RonR JoeinForum and everyone else who help with this. I'll ask a couple of questions and then will write a how-to for dummies for the next person who, like me might have little background in this.  First, (selfishly) my questions:  

Everything is working, (plus as I will describe I've added a sip client and edited the DigitMap accordingly).  However, the following issues remain:

1.   with Skype, when I can call out everything is fine, however, incoming calls trigger siptosis which seems to see the call and then immediately disconnects after ringing once on the computer and never ringing on the OBIHAI phone.  Any ideas why this may be?  Attached is a screenshot showing a)  A typical error I get when executing the bat file (shown on the top line).  b)  The log of when calling into Skype.  any diagnosis of this would be appreciated!!

EDIT:  forum uploader is full so screenshot not attached

2.   When calling out from the OBIHAI phone via Skype, sometimes the call recipient cannot hear me.  If I push a key on the dial pad generating a tone they can hear me following that.  Does this make sense?  Is there something I can do to overcome this?  Is there something wrong with my configuration?  Possibly related, if an OBIHAI Skype call is underway I often hear it both on my hard phone and on the Skype computer phone.  If I push the keypad on the computer I also hear that tone.  Is it normal to have sounds across both the computer and the OBIHAI phone?

3.   I don't understand the DigitMap phraseology and just copied what RonR posted.  I observed the following:

  a.   the references for service providers beyond number two were removed so when I added a sip service provider as SA3 I reentered this manually.

  b.   He uses "#1...#" to trigger service provider 2, where OBIHAI's default language uses **2, I haven't done it yet but will likely change this back so everything is consistent.

I think that's it for questions/comments other than THANK YOU GUYS VERY VERY MUCH!

Now I'll post my how-to for dummies in the next post...