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> #.
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.
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.
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.
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.
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.
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
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.
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.
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#).
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
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}
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
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.
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.
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!
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
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
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.
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...
Quote from: ccclapp on August 16, 2012, 07:53:08 PM
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?
I've never been able to have a conversation with two different Skype clients registered with the same account. Although I realize that you can have several different computers running Skype with the same login credentials, I thought only one client could connect to the call. In your case you more than likely have one computer running two instances of Skype using the same acct (siptosis and Skype). This is more than likely the reason it seems to crash.
It seems that you could can occasionally use the OBi hard phone and your Skype computer phone on the same call. I think you should consider yourself lucky that you are able to do that at all. That is basically a form of conference call. To have a problem free conference call I think setting the Skype on the computer under a different Sykpe name would be more reliable.
Can't wait to see that How to article. Can I suggest that you start a new thread for this, with a reference to this thread. It would be nice to have a good "How to" in the title so that it will be easier to find in the future.
thanks QBZappy
Just to clarify, I'm not doing what you imagine. On the local computer I'm running a single instance of Skype with siptosis loaded. Then, I'm calling that Skype number from my iPhone Skype app login to a different account. Thus none of that should be a factor.
As to my being able to hear a call from my OBIHAI phone both on that phone and on the adjacent computer (which is running Skype and siptosis) it doesn't go so far as to be a conference feature, more of an annoyance because sometimes sounds come out of both. I would want sound only coming from the OBIHAI phone with a computer remaining silent.
As to my how-to, as you suggest a new post may be worthwhile but there is still very good information in this post so linking back to it and giving all the credit to RonR and others in putting yourself may be the cleanest solution...
ccclapp,
Quote from: ccclapp on August 16, 2012, 08:53:10 PM
As to my being able to hear a call from my OBIHAI phone both on that phone and on the adjacent computer (which is running Skype and siptosis) it doesn't go so far as to be a conference feature, more of an annoyance because sometimes sounds come out of both. I would want sound only coming from the OBIHAI phone with a computer remaining silent.
It looks like others before you have had this issue. Short of turning off the mic and speakers, it may not be possible to control that behaviour. Read the entire thread. The info there might interest you.
Re: Using # in DigitMap for routing to SIPToSis gateway is not working.
http://www.obitalk.com/forum/index.php?topic=701.msg4307#msg4307
Thank you RonR i got this working nicely :).
Just one thing to add is in this part:
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
While in
SkypeToSipAuth.props you need to comment this line (add a # before the line):
*,play:clips/invalidDest.wav
So the final
SkypeToSipAuth.props file should look like this:
#Default: all incoming skype callers get the invalid destination message
#*,play:clips/invalidDest.wav
*,sip:siptosis@YourObiIPAddress:5061,YourSkypeUserID
NOTE: I didn't have to setup my antivirus firewall or port forward other ports.
If any one know a way to make this work on an Android smartphone would be great, the .bat must be running in the background for this to work, maybe the OBI guys can integrate this to the OBIon app? well they haven't updated their Android app since 2011 so ::)... sigh!
Currently im using Groove IP in my smartphone for my google voice account which is the only i had setup in my Obi110 anyway.
~ V
Quote from: Veronica on August 17, 2012, 12:15:25 PM
Thank you RonR i got this working nicely :).
Just one thing to add is in this part:
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
While in SkypeToSipAuth.props you need to comment this line (add a # before the line):
*,play:clips/invalidDest.wav
So the final SkypeToSipAuth.props file should look like this:
#Default: all incoming skype callers get the invalid destination message
#*,play:clips/invalidDest.wav
*,sip:siptosis@YourObiIPAddress:5061,YourSkypeUserID
hi,
I'm in the midst of writing a idiot's guide for this and it meant I am a case in point idiot. Would you mind elaborating on your recommendation above? I thought using the #is simply a way of putting in text that would not control anything. I must be wrong because you seem to be generating commands with your lines which start with #.
that point aside, would you please describe what you are text is doing in terms of commands?
Thanks
Quote from: ccclapp on August 17, 2012, 12:21:44 PM
Quote from: Veronica on August 17, 2012, 12:15:25 PM
Thank you RonR i got this working nicely :).
Just one thing to add is in this part:
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
While in SkypeToSipAuth.props you need to comment this line (add a # before the line):
*,play:clips/invalidDest.wav
So the final SkypeToSipAuth.props file should look like this:
#Default: all incoming skype callers get the invalid destination message
#*,play:clips/invalidDest.wav
*,sip:siptosis@YourObiIPAddress:5061,YourSkypeUserID
hi,
I'm in the midst of writing a idiot's guide for this and it meant I am a case in point idiot. Would you mind elaborating on your recommendation above? I thought using the #is simply a way of putting in text that would not control anything. I must be wrong because you seem to be generating commands with your lines which start with #.
that point aside, would you please describe what you are text is doing in terms of commands?
Thanks
Hmmm ok i don't think i really understand what you mean but i'll reply you anyways.
When the main program calls a file for an instruction it will only read code that is uncommented that means code without
# in front.
# is used to comment text like instructions that serve as a guide for the user/developer and also for comment code that you don't want to be detectable, and so "readable" by the main program.
Now this specific command:
*,sip:siptosis@YourObiIPAddress:5061,YourSkypeUserID
is used when one of your Skype Contacts calls you thru Skype off course. It will redirect the current Skype call to your Obi thru port 5061 which belongs to SP2 (As per RonR instructions)
That's how i understand it and assimilate it.
~ V
hi Veronica
as to the first part of my post, I believe you understood it. If I read correctly you have confirmed what I thought: any text beginning with a "#" is simply text to be used as reminders notes etc. and will not trigger any commands.
That being the case what I don't understand is your post that: "the final SkypeToSipAuth.props file should look like this:...
...#*,play:clips/invalidDest.wav"
because that line has a #at the beginning it doesn't do anything, does it?
If that is the case, it doesn't look to me like you changed anything. I'm just trying to be sure I understand if you've added/changed anything from what RonR posted because as I say I'm trying to write this up in explicit detail so dummy users like me know how to configure OBIHAI for Skype
Thanks
Original file after RonR instructions will look like this:
#Default: all incoming skype callers get the invalid destination message
*,play:clips/invalidDest.wav
*,sip:siptosis@YourObiIPAddress:5061,YourSkypeUserID
I just added one more step to do in that file which consist in comment (#) *,play:clips/invalidDest.wav line, because if you don't when your skype contacts try to call you they will get invalid destination message so it won't ring your phone plugged to the OBI.
If you DO comment that line then the main program won't detect it but it will detect *,sip:siptosis@YourObiIPAddress:5061,YourSkypeUserID which is the one we want to be detected, then your phone will ring successfully and you will be able to speak with your Skype Contacts.
~ V
thanks again!
Just being doubly sure I understand:
Are you saying that RonR's line "*,play:clips/invalidDest.wav" was intended to be an internal note and not a command and that it should have had a #in front of it( so it would be a note and not a command) and you are correcting that?
If that's true, great call. I never would have figured that out. Note: in following Ron's guide I can make outgoing Skype calls on my OBIHAI, but incoming calls to my Skype address fail. Possibly you're correction will fix this...
Quote from: ccclapp on August 17, 2012, 01:15:02 PM
thanks again!
Just being doubly sure I understand:
Are you saying that RonR's line "*,play:clips/invalidDest.wav" was intended to be an internal note and not a command and that it should have had a #in front of it( so it would be a note and not a command) and you are correcting that?
If that's true, great call. I never would have figured that out. Note: in following Ron's guide I can make outgoing Skype calls on my OBIHAI, but incoming calls to my Skype address fail. Possibly you're correction will fix this...
Yes, will fix it if you followed Ron's instructions to the T.
I don't know if RonR is the creator of the SipToSis package, by default
*,play:clips/invalidDest.wav is un-commented but it must be commented if you want this to work. RonR gave instructions but he forgot to mention to comment that line, after that incoming calls will work, the whole method will work 100%.
~ V
Veronica
QuoteYes my suggestion will fix it if you followed RonR instructions to the T.
fabulous!! This solved my problem (at least the one having to do with Skype and OBIHAI)
As previously stated in following Ron's guide I was able to call out from my OBIHAI phone via Skype but all incoming calls to my Skype number failed. Your correction solved this!
Many thanks
I've successfully worked through the Skype set up with my OBIHAI but because of my more limited telephony experience it took a while to understand RonR's excellent instructions. To help other less experienced users I've written a detailed step-by-step guide that follows what Ron wrote but with more explanation...
http://www.obitalk.com/forum/index.php?topic=3920.0
I'm only doing this to be helpful not to try to take any credit for RonR's work.
If anyone sees any errors please let me know.
Thanks for everyone's help!
Hello everybody, i decided to make a modded package for this Set-Up that Ron-R kindly shared...
*It is meant for Windows users
*This is an on-the-go package with all the instructions so you don't need to check back the forum for the entire instructions.
1. Extract the package as extract here.. you will then see a SipToSys folder
2. Read:
- Readme-INSTALLATION.txt
- Readme-OBI Configuration.txt
Download Links:
SipToSis-for-OBI+Win.zip
MediaFire (http://"http://www.mediafire.com/?yiyczohlw856osn") - GDrive (http://"https://docs.google.com/open?id=0B3XeIVyUffMKVGlvRDNub3RSRDg") - Dropbox (http://"https://dl.dropbox.com/u/21630798/SipToSis-for-OBI%2BWin.zip")
~ V
Hi.
Veronica,
were you able to find a solution for the smartphone, android?
I have it working in Windows and Linux but I can't seem to get it working on my Galx S2 or MK802.
Thanks all for your help.
I have installed sipTosis and Java 1.7 on an XP box. When I go to run sipTosis_win.bat, the batch file finds the Java path OK, but instead of running the java script the batch file returns the java command line help and then exits. Any ideas what I am missing?