CSipSimple + Sip2Sip + Obi202 + Google Voice

[threehappypenguins]

I live in Canada and currently have an Obi202 with AcroVoice on SP1 and Google Voice on SP2 (no GV phone number). AcroVoice uses autoprovisioning so I set up GV on the Obi without adding the Obi to the dashboard through Obitalk. From what I have read, it is possible to use CSipSimple on my Android, register it with Sip2Sip (which I have done), and then somehow configure Sip2Sip on my Obi (which I have not figured out... I assume it must be configured on SP3?). I read a whole bunch of posts that this can be done, but nobody explains HOW. It is all Greek to me! Then I want to use CSipSimple to call my Obi via Sip2Sip and reroute the call so it either goes through AcroVoice or Google Voice. I also want to set it up so that incoming calls will be rerouted to go to my Android and ring on there (when I want them to... for travelling purposes). I already have CSipSimple and registered my Sip2Sip account with it.

I tried using ObiON, but I do not understand how to get it to work without using Obitalk, which will conflict with AcroVoice provisioning.

Can this be done? If so, how? Just so you know, I have NO IDEA how to configure the Obi. In order for me to get Google Voice onto the Obi without adding it to the dashboard, I had to first add the Obi device to the Obitalk dashboard, added Google Voice to SP2, then went into expert configuration and copy and pasted the changes that it made to it into the configuration via the Obi's IP address, and then deleted the Obi off the dashboard.

So if someone could direct me what to do and where to put the information, that would be great!!!

[ianobi]

threehappypenguins,

This may take a few days, as I'm not here every day. Hopefully, others may dive in and help when I'm not around.

With a few changes for the OBi202, this post covers most of what is needed:

http://www.obitalk.com/forum/index.php?topic=4647.0

I know some of it looks daunting, if you are not used to OBi stuff. In your case you can do all the changes in the OBi from the web page at its IP address. To make each change: Uncheck the "default" box and leave it unchecked, make all required changes on that page, click on submit at the bottom of the page, then click reboot at top of page. Wait two minutes for OBi to reboot and settle down. Refresh page.

Before you start, you will need the following:
Your router must have a fixed IP Address or a dns type address (penguin@happyhome.dns.com) for incoming calls. For outgoing calls you will need another SIP provider, may as well be sip2sip, set up on sp3. You need to set the CSipSimple filters as described in the above post.

For this post let's get the outgoing from OBi working:

Sign up for another sip2sip account. Then log into OBi web page:

Service Providers > ITSP Profile C > SIP
ProxyServer: sip2sip.info
ProxyServerPort: 5060 (default)
OutboundProxy: proxy.sipthor.net
OutboundProxyPort: 5060 (default)
Submit/reboot/wait 2 mins

Voice Services > SP3 Service
Enable: checked
X_ServProvProfile: C
X_RingProfile: A
X_CodecProfile: A
X_InboundCallRoute: (we will change this later when I've worked it out!)
X_RegisterEnable: checked
AuthUserName: yoursip2sipusername
AuthPassword: your sip2sippassword
CallerIDName: hpenguin (whatever!)
MaxSessions: 4
Submit/reboot/wait 2 mins

System > Status should now show sp3 registered.


Voice Services > SP1 Service > X_InboundCallRoute:
Delete: ph,ph2
Insert: {sp3(userid@sip2sip.info;ui=$1),ph,ph2}
Submit/reboot/wait 2 mins

Where userid is the sip2sip userid of the account registered with CSipSimple. This will "fork" incoming calls on sp1 to the android phone and the OBi phone. First device to answer gets the call. If you want the calls to be either OBi or android rather than both, we can sort that out later.

Well that's calls from OBi to your cell phone sorted! Testing is best done while the cell phone has a wifi connection.

To call direct from OBi phone to android, you can set up a speed dial entry: sp3(userid@sip2sip.info;ui=$1)

User Settings > Speed Dials > any spare slot. Submit/reboot etc.

Next post we will sort cell phone to OBi.

Edit: Changed to: X_ServProvProfile: C

[threehappypenguins]

Thank-you ianobi!!!

I implemented most of what you said. Except for under Voice Services > SP3 Service I changed the X_ServProvProfile: A to C. Then I was able to register the additional sip2sip account.

I actually technically do not have AcroVoice as of yet, but I set it up similarly with my in-laws first, so I know what it looks like when it is functioning with Google Voice. I am in the middle of porting my number now. my in-laws have a Google Voice number, so what I did was replace my Google Voice username and password in sp2, and then in the sp2 X_InboundCallRoute I put what you suggested (except that I replaced userid with my username, of course).

I made sure that I had CSipSimple open on my Android (via WiFi) and registered with sip2sip under my original username (the first one). I then used my current landline (that is still in service since my number has not been ported yet), and called my in-laws Google Voice number. The Obi rang, but my Android did not ring. I checked the settings in CSipSimple and made sure that the Use WiFi was checked for incoming calls. I even checked under User Interface and saw that the ringtone was silent, so I picked a ring tone. Still no incoming call.

I am at a loss. Can you see what I am doing wrong?

[ianobi]

I will edit my first post in case others wish to copy it:
X_ServProvProfile: A to C. I'm extrapolating from OBi110 and OBi documents are a bit lacking in this area again  

In the following two uses userid must be the userid of the sip2sip account registered on CSipSimple:

Edit: drop the ui=$1 sip2sip does not like it. Further research needed. So use:

Voice Services > SPX  Service > X_InboundCallRoute:
Delete: ph,ph2
Insert: {sp3(userid@sip2sip.info),ph,ph2}
(SPX should be the incoming service)

To call direct from OBi phone to android, you can set up a speed dial entry: sp3(userid@sip2sip.info)

I would suggest setting up the speed dial for testing. If you put sp3(userid@sip2sip.info) in speed dial slot 10, then do the usual submit/reboot etc. Then dialling 10# from the OBi phone should call the sip2sip account registered on the cell phone.

After a call attempt, such as your example calling into GV on sp2, have a look in Status > Call History to see what happened. It should show the call going to ph, ph2 and forking to sip2sip.

Next post really will deal with cell phone to OBi  

[ianobi]

Cell phone to OBi. This assumes that:
CSipSimple is registered with a sip2sip account.
The two filters for that account are in place (Prefix all with **7, suffix all with @happypenguin.com:5062 or fixed IP address:5062).
CSipSimple is set to "dialer intergration" checked.

5062 is the default UserAgentPort of sp3 in the OBi. You may need to port forward 5062 in your router to make calls incoming to the OBi from outside your local subnet.


I'm assuming that your PrimaryLine is set as follows:

Physical Interfaces -> PHONE Port -> PrimaryLine : SP1 Service

In your case this will be Acrovoice. If that is true, then this should work:

Voice Services -> SP3 Service -> X_InboundCallRoute (SP3 must be configured for SIP):

{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**3:>(Msp3)),(Mcot)>(<**3:>(Msp3)):sp3},{(Mcot)>(<**7**4:>(Msp4)),(Mcot)>(<**4:>(Msp4)):sp4},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Msp1)),(Mcot)>(Msp1):sp1},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph,ph2},{ph,ph2}

cot is a User Defined Digit Map:

(12345678|87654321|11223344)

My cot happens to have three Caller IDs in it. Using this method means you only have to change cot if you add or change Caller IDs, rather than change every reference of Mcot in the InboundCallRoute. cot has to contain the sip2sip user name of the account set up on CSipSimple.

If your PrimaryLine is set as follows:

Physical Interfaces -> PHONE Port -> PrimaryLine : SP2 Service

In your case this is GV. Then change:

{(Mcot)>(<**7:>(Msp1)),(Mcot)>(Msp1):sp1}

to:

{(Mcot)>(<**7:>(Msp2)),(Mcot)>(Msp2):sp2}

In all cases with DigitMaps above, if you cut and paste from here, be very careful not to import any spaces or miss bits on the end of lines. (It happens!)

Both the incoming and outgoing setups can be refined to achieve a few different things, but we can look at that after we get all the basics working. Don't be surprised if things don't work first time. Good luck  

[azrobert]

said by ianobi:

Voice Services > SP1 Service > X_InboundCallRoute:
Delete: ph,ph2
Insert: {sp3(userid@sip2sip.info;ui=$1),ph,ph2}

then

Edit: drop the ui=$1 sip2sip does not like it. Further research needed.

I am also researching why ui=$1 doesn't work with SIP2SIP.
I did come up with 2 solutions that provide CallerID.

Solution 1
Get a free account at PBXes.org.
Define an extension.
Under Device Option change the Dial option to "SIP/userid@sip2sip.info"
Define an InboundRoute to route all calls to this extension.
Change OBi X_InboundCallRoute to {spx(userid-exten@pbxes.org;ui=$1),ph}
Downside: Adds another service provider.

Solution 2
Change OBi X_InboundCallRoute to the default.
Get a free IPKall phone number.
Config IPKall to SIP URI = userid@xx.xx.xx.xx:50xx  (your home external IP address and OBi port#)
Add the IPKall number to your GoogleVoice Settings.
After you verify the IPkall number to GoogleVoice change the IPKall definition to URI = userid@sip2sip.info
In GoogleVoice Settings forward all calls to GoogleChat and IPKall
Downside: IPKall sometimes does not provide the correct CallerID.

Edit: You can skip pointing IPKall to the OBi and verify it on your Android handset.

[ianobi]

@ azrobert

Yes, it's odd that sip2sip does not seem to allow the ui=$1. I chose sip2sip originally because (a) its free and (b) has a very good debug facility. I may try another provider.

I would like to keep the number of accounts / providers to a minimum. On that topic I have a few more comments, but I'll go back to the original thread:

http://www.obitalk.com/forum/index.php?topic=4647.0

[threehappypenguins]

It worked!!! At least the calling TO my Android part!!! I was confused when you said to set up another sip2sip account. That threw me off, but I just assumed that I wasn't picturing the whole process properly, and just went with what I thought you were saying. So then I used my original sip2sip account, and disregarded the second one I created. That, and getting rid of the ;ui=$1 did the trick. I was able to call the Google Voice number (borrowing my in-law's account for testing purposes), and it rang on the Obi and the Android at the same time, like you said it would. Only the caller ID that shows is always the one that I set up to be my CSipSimple caller ID. Also, I checked Status > Call History and saw it successfully being forked to the sp3 and to the Sip2Sip account from there.

When I tried to implement the rest of your suggestions, there were several issues:

1. Speed dial will not work. I get the reorder tone (fast busy signal) when I dial 10#. I am not sure if this has to do with anything, but Trevor from AcroVoice knows my set up for Google Voice and set up the DigitMap and OutboundCallRoute for me so that when I dial a local number I only need 7 digits (we only have one area code out here in Nova Scotia), and when I dial long distance, I dial 1+area code+7 digits. And the primary line was (and normally will be) SP1, but for testing purposes since I don't have AcroVoice set up yet, I changed it and set up the Primary Line to SP2.

Under Physical Interfaces > PHONE1 Port
DigitMap: (!**5S0|[2-9]11S0|1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|0|011xx.|*xx|***|#)
OutboundCallRoute: {(<#:>):li},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{([2-9]11S0|[2-9]xxxxxxS0|011xx.|0|*xx):sp1},{(1[2-9]xx[2-9]xxxxxx):sp2}

2. Adding the prefix and suffix in CSipSimple was confusing. I Googled how to do this, and I think I figured it out. What I did was:

Settings > Filters > Sip2Sip > Add Filter / rewrite rule
Changed Can't Call to Rewrite
Changed Starts with to All
Changed Replace match by to Prefix by (and then Suffix with when making the separate suffix rule)

The other thing that confused me is the Prefix rule that you wanted me to create. What exactly am I putting in there? You said:

suffix all with @happypenguin.com:5062 or fixed IP address:5062

I tried putting in my Obi's ip address (is that what you want? The Obi202 address that I made static?). So for example: 192.168.x.xx:5062
As for the mentioning of @happypenguin.com, do you mean I am putting in my User ID for sip2sip? So if my user ID for sip2sip is represented by userid, then it would be @userid.com? I am confused.

3. The port forwarding threw me off too. I don't know how to do port forwarding, nor do I understand it. I mean, I think I understand why you want me to do port forwarding. Is it so that I can make a call from my Android to my Obi while my Android is connected to a separate Wifi than the home network my Obi202 is connected to?

I went to Router Configuration > Port Forwarding and then stared blankly at it for a while. I considered for a moment to choose Both under Protocol, and then type 5062 under StartingPort and 5062 under EndingPort thinking that this would cover all bases... but then I looked at ServerIPAddress and thought "What Server? What do I put in here?" It had a spot in it to put in the last one or two number(s) of an IP address. Plus, I didn't know what RuleDescription was for, and whether it was just simply a label for the port forward thingy that I am supposed to create or if it was actually some sort of code or something. So I just left the port forwarding issue alone for now. I am not sure if that affected anything since currently I am testing while connected to my home network on my android.

4. I did not understand you with all that Mcot stuff. You said that a cot is a User defined Digit Map. To tell you the truth... I don't even know what a Digit Map is either. All I know about it is if you put some stuff in the DigitMap part of PHONE1 Port, then it tells the phone what will happen when I dial certain digits or amounts of digits. So... it is a map for dialing. That is to the extent of which I understand. So when you say that your cot has 3 caller ID's in it, I have no idea what you are talking about... so I took a stab at what you were saying, and changed the first Mcot to say (assuming that my sip2sip user name is represented by userid):

Muserid

So I ented under X_InboundCallRoute under SP3:

{(Muserid)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**3:>(Msp3)),(Mcot)>(<**3:>(Msp3)):sp3},{(Mcot)>(<**7**4:>(Msp4)),(Mcot)>(<**4:>(Msp4)):sp4},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Msp2)),(Mcot)>(Msp2):sp2},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph,ph2},{ph,ph2}

That didn't work (404/user not found was the error when I dialed with the native dialer, choosing Sip2Sip), and I didn't understand what you were talking about with the Caller ID's, so I changed every time it said "cot" to "userid" in the whole big line of confusingness that you pasted   and that did not work either.

[ianobi]

OK, there was a lot of jargon, numbers and general OBi type stuff to deal with. Be patient, we will get there. It's late here in my time zone, but tomorrow I will try to be less confusing.

I could not live with myself if I turned threehappypenguins into threesadpenguins 

[ianobi]

Firstly, an apology. This subject has been changing as we have been posting, so there has been some confusion. Secondly, please bear with me. On this forum we are all amateur OBi users like yourself just helping each other out, we do not work for Obihai. This is a big subject for someone new to OBi – you are diving into the deep end. Hey, penguins are good swimmers, you should be ok 

The principles:

Calls from OBi to CSipSimple via sip2sip, leave OBi as:
aaaaaaaa@sip2sip.info
Where aaaaaaaa is your sip2sip username.
sp3(aaaaaaaa@sip2sip.info) tells the OBi to send the call out via sp3.

Calls from CSipSimple to OBi via sip2sip, leave CSipSimple as:
**7xxxxxxxxxxx@yourpublicipaddress:5062
Where **7 is for routing in OBi. xxxxxxxxxxx is the number dialled. @yourpublicipaddress is your router public WAN address. (This may be in format xx.xxx.xxx.xxx. If this is likely to change you may consider using a free dynamic dns provider to get a fixed address something like @penguin.ddns.com). 5062 is the port number associated with sp3 in the OBi, so it tells the router to send the call there.

On arrival at your router, the router will remove the @yourpublicaddress and use 5062 to route the call to sp3 InboundCallRoute. OBi will remove the **7 and route the call according to the InboundCallRoute rules.


CSipSimple:

Assumes a sip2sip account set up already.

Settings > User Interface > Dialer integration : check
Settings > User Interface > Call logs integration : check

Settings > Filters > sip2sip > Add filter / rewrite rule > Rewrite > All > Prefix by : **7
Settings > Filters > sip2sip > Add filter / rewrite rule > Rewrite > All > Suffix with : @yourpublicipaddress:5062

Yes, I agree filters are a bit hard to get the hang of!

Now, using the native android dial pad (not the CSipSimple dial pad), when you select a contact or input digits, you will be given a choice to send using mobile or sip2sip.


Router:

Obihai recommends that you port forward the following ports:
Allow Outgoing:
TCP Ports: 6800, 5222, 5223
UDP Ports: 5060 to 5063, 10000 to 11000, 16600 to 16998, 19305
Allow Incoming on UDP Port: 10000

I'm sorry I'm not much of a router expert. Under "Open Port" mine has fields "Start Port", "End Port" , "Local Host". "Local Host" is the OBi address such as 192.168.x.xx


OBi202:

We do not need the sip2sip account set up on the OBi. Please remove those settings and replace with the following:

Service Providers -> ITSP Profile C -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile C -> SIP -> X_SpoofCallerID : checked

Voice Services -> SP3 Service -> Enable : (checked)
Voice Services -> SP3 Service -> AuthUserName : (any letters or numbers)
Voice Services -> SP3 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP3 Service -> X_ServProvProfile : C
Voice Services -> SP3 Service -> X_UserAgentPort : 5062
Voice Services -> SP3 Service -> CallerIDName : Whatever
Voice Services -> SP3 Service -> MaxSessions : 4

Voice Services -> SP3 Service -> X_InboundCallRoute (Assumes Primary Line is set to sp2):

{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**3:>(Msp3)),(Mcot)>(<**3:>(Msp3)):sp3},{(Mcot)>(<**7**4:>(Msp4)),(Mcot)>(<**4:>(Msp4)):sp4},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Msp2)),(Mcot)>(Msp2):sp2},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph,ph2},{ph,ph2}

User Settings > User Defined Digit Maps > User Defined Digit Map2 > Label : cot
User Settings > User Defined Digit Maps > User Defined Digit Map2 > DigitMap : (aaaaaaaa)

Where aaaaaaaa is your sip2sip username.

No need to change any "Mcot"s, OBi will substitute your sip2sip username by using the cot DigitMap.

Dialling from your android phone native key pad, selecting the sip2sip option to send:

**1xxxxxxxxxxx > Acrovoice on sp1
xxxxxxxxxxx > GV on sp2 (no need for ** code as this is your Primary Line)
**0 > Obi auto attendant
0 > Rings both OBI phones

I'm not sure why your speed dial did not work. No digit maps etc are involved with this format:
sp3(aaaaaaaa@sip2sip.info)

It is a lot of information all in one go. Feel free to come back for more.

[threehappypenguins]

LOL!!! YOU will not make me threesadpenguins! Obihai is on the verge of making me threesadpenguins because there is no simple work around for a person like me that just wants a simple workaround to make cheap calls on my Android phone via WiFi (and Talkatone and GrooveIP SUCK!!!). We have not had a cell phone for years because they are too expensive and we are a one income family (3-year-old and 10-month-old to take care of and home school!!!) So my husband and I finally bought one with cash and are just buying a simple $100 card and are going to make it last a year so that our cell phone will be $9.58 /mo on average after taxes. So I really REALLY appreciate your help because I was not able to get ANYWHERE with Googling for answers! I realize that I am diving in the deep end, but I really do not have much of a choice if we are going to live cheaply and comfortably.

I seem to be making progress. I can now call my Obi phone from the Android. But it is completely accidental... so I am confused (again... lol!). *Scratch that... Now it is not working. I have been typing this reply for HOURS on and off (going between this and watching/feeding/taking care of children).... ARGGGG... I didn't change ANYTHING and now it is not working!!!* Let me start from the beginning.

When you originally told me to port forward 5062, you said:

5062 is the default UserAgentPort of sp3 in the OBi. You may need to port forward 5062 in your router to make calls incoming to the OBi from outside your local subnet.

I thought when you said "in your router" you meant the Obi. Are you talking about my internet router? If so, it gave me several settings. It is a Motorola SBG6580. I went to Advanced > Forwarding and then clicked on the Create IPv4 button.

I then get several options. Under Local IP, I typed my local IP address (I found it by opening command prompt, typing in ipconfig and finding the IPv4 address). Then in Local Start Port & Local End Port I typed in 5062 in both boxes. Then in External IP, I typed my public IP address which I found by typing public ip address into Google. Then in External Start Port & External End Port I also typed 5062. For protocol, I did not know what to choose (whether TCP or UDP) so I chose Both. I left the Description blank, and I chose On for Enabled. Then I clicked the Apply button. Did any of it make any sense to me? Nope! So if someone is reading this that knows how to port forward, please clarify for me!

You said:

Router:

Obihai recommends that you port forward the following ports:
Allow Outgoing:
TCP Ports: 6800, 5222, 5223
UDP Ports: 5060 to 5063, 10000 to 11000, 16600 to 16998, 19305
Allow Incoming on UDP Port: 10000

I'm sorry I'm not much of a router expert. Under "Open Port" mine has fields "Start Port", "End Port" , "Local Host". "Local Host" is the OBi address such as 192.168.x.xx

So again, you are talking about port forwarding in the internet router, and not the Obi? It just throws me off because in the Obi menu there is also an option to do port forwarding under Router Configuration > Port Forwarding

With the above suggestion for port forwarding, I do not even know where to start. I do not have an option to allow incoming. Only the settings options that I explained when I was trying to port forward 5062.

Furthermore, from what I understand, Obihai said in this FAQ page http://www.obihai.com/docs/OBiFAQ.html:

TCP/UDP Ports your Firewall should not block
In order for your OBi to be able to send packets w/o interruption, please allow the following ports for outgoing:
TCP: 6800, 5222, 5223
UDP: 5060, 5061, 16600 to 16998
Allow Incoming on 10000

It does not say anything about port forwarding, but it is about not allowing your Firewall to block the above said ports. So, not knowing what to do or where to start, I left that all alone.

Dialer integration and call log are checked in CSipSimple. I signed up for a dynamic dns (I think? I don't know...) from https://www.no-ip.com. Assuming userid is my username, my dynamic dns (at least I think it is dynamic... whatever that means...) is userid.no-ip.org. So I put in the suffix part of CSipSimple @userid.no-ip.org:5062. I tested this in command prompt by typing ping userid.no-ip.org and it worked just fine.

I made these changes that you suggested here

Voice Services -> SP3 Service -> X_InboundCallRoute (Assumes Primary Line is set to sp2):

{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**3:>(Msp3)),(Mcot)>(<**3:>(Msp3)):sp3},{(Mcot)>(<**7**4:>(Msp4)),(Mcot)>(<**4:>(Msp4)):sp4},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Msp2)),(Mcot)>(Msp2):sp2},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph,ph2},{ph,ph2}

User Settings > User Defined Digit Maps > User Defined Digit Map2 > Label : cot
User Settings > User Defined Digit Maps > User Defined Digit Map2 > DigitMap : (aaaaaaaa)

Where aaaaaaaa is your sip2sip username.

I also made these changes here

We do not need the sip2sip account set up on the OBi. Please remove those settings and replace with the following:

Service Providers -> ITSP Profile C -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile C -> SIP -> X_SpoofCallerID : checked

Voice Services -> SP3 Service -> Enable : (checked)
Voice Services -> SP3 Service -> AuthUserName : (any letters or numbers)
Voice Services -> SP3 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP3 Service -> X_ServProvProfile : C
Voice Services -> SP3 Service -> X_UserAgentPort : 5062
Voice Services -> SP3 Service -> CallerIDName : Whatever
Voice Services -> SP3 Service -> MaxSessions : 4

But then had to undo them because it did not work.

After undid the last noted changes and put the sip2sip account back on the Obi, I was able to make calls from the Android to the Obi. Only it did not matter who I dialed. It is like it ignored what I dialed, and just always called the Obi. I do not know why.

I also attempted to call the GV number from my home phone ptsn line to test it forking to the Android, but it did not work anymore. So I started poking around and noticed that the only thing I did not undo from the changes you told me to make (because I forgot about it) was Service Providers -> ITSP Profile C -> SIP -> X_SpoofCallerID : checked. So I unchecked it, and I was able to call the Android again.

As noted wayyyyy above in this reply, the setup stopped working. I cannot make outgoing calls on my Android. It just... stopped. I pinged userid.no-ip.org again in cmd, and at first it did not work. So I logged back in no-ip.org, clicked on "modify" did nothing at all, clicked "update" and magically it started pinging again from cmd. I tried dialing out from my Android... and... nothing.

ARGGGGG

Oh, and my eyeballs hurt.

[threehappypenguins]

It is 3pm here in Halifax and I need to start cooking supper and do some laundry, so that will be a nice getaway. Oh the joys of networking/telecommunications...

[ianobi]

Sorry to hear that you are having problems. Each time I mentioned router I did mean your internet router. The info in the OBihai faq was not updated when they brought out the OBi202 - for 5060, 5061, read 5060 to 5063.

If you would like a different way forward, then you could go with Obion on Android. It does not look to me that Acrovoice are doing actual "provisioning", rather they have been helpful in giving configuration details. If you were to note those carefully in a word doc along with the GV details, then you could copy and paste them back later. This way you could do a factory reset and start again using the OBiTALK portal and OBiON. You should read the OBiON for Android section first on this forum, most users are not very impressed. The CSipSimple method is quite complex, but it really does work! Several people on this forum use it.

Whichever way you choose to go, there are helpful people on this forum who will support you.

[QBZappy]

Hi folks,

Trev (forum member) is the public face behind Acrovoice (http://www.acrovoice.ca/). He is very active over at DSL Reports (http://www.dslreports.com/). He can also be reached using the PM feature of this forum as well. I have seen him give configuration advice to his customers. If you are a paying customer of Acrovoice you should be entitled to get support from them. After all that is the reason you are paying them every month for the service. From what I can see their pricing reflects an offering of a premium service. If you go that route post back with the solution or a link where it can be found.

You won't need to bother with Ian's English accent. (Of course unless you like it). 

PS
I think Ian is bucking for a sainthood. (or knighthood in his case).

[ianobi]

Nice input QBZ. You are indeed a jolly good chap, despite what the others say about you 

[threehappypenguins]

Ahhh yes. Trev. He has been incredibly helpful while I was setting up my in-laws Obi. He blocked their long distance in the middle-of-the-road plan so I could get GV on and they get long distance for free. Trevor has gone above and beyond with help, and I have emailed him so many times that I cannot count! So I did not want to drive him crazy with another email... LOL!

Obion will not appear to work. I originally had my inlaws Obi110 on the Obitalk dashboard, and it was interfering with the AcroVoice provisioning. We tried working around it, but Trevor eventually had to have me completely disable it and delete it off of the dashboard. Then everything worked fine (well, except for some unrelated issues...). Now that I have my own Obi (Obi202), I have put it on the dashboard and attempted to simply use Google Voice with Obion. I do not understand how Obion works and why it will not work right. It doesnt matter what I dial, but it always calls the Obi, and not the number that I was dialing. So I gave up. And like I said, Obitalk interferes with AcroVoice provisioning, so I have no use for Obion anyway. I tried to play around with it to see if I could have on the dashboard, use Obion with Obitalk, and yet have the Obitalk disassociated with the provisioning via the Obi IP address, and I just could not see how it is possible. If any of that made any sense...

Thanks for trying Ianobi!!! 

I guess I will have to play around with this more... somehow. maybe read a nice 1000 page book on ports, forwarding, networking, telecommunications, etc. Anyone want to babysit for a week straight? 

[threehappypenguins]

I started this thread http://www.obitalk.com/forum/index.php?topic=4726.0 because the outgoing call was only going intermittently (sometimes would go through, sometimes would time out), and I figured it was a port forwarding problem. I think that setting dmz on my router (not the Obi) solved the problem. I still have yet to test the calls on a different network. Right now my Android is on the same WiFi network so we will see when it is connected elsewhere outside my home network.

The last problem I was having was the Android ALWAYS called the Obi. I could dial any number I wanted, it didn't matter. It would just go to the Obi. It was not recognizing my sip2sip username as a trusted number and so it was always being sent to the Obi. ianobi, you are the best! He helped me via messaging on this forum.

OBiAdminGuide:

literals [in DigitMaps] - Any combination of 0-9,*,#,+,-,A-Z,a-z, except m, M, s, S, x, X which have special meaning in the digit map syntax.

http://www.obihai.com/OBiAdminGuide.htm

RonR:

The fact that you're reaching the PHONE Port means you're almost there.

The problem is likely that you don't have the username from your IP Phone in the cot User Defined DigitMap.  Look at the OBi Call History and make sure you're using the correct username.

Also, if there are any reserved characters (m, M, s, S, x, X) in the username, they have to be surrounded by single quotes.  For example, if the username is xlite, it would have to be entered in the cot User Defined DigitMap as : ('x'lite)

http://www.obitalk.com/forum/index.php?topic=2454.0

It turns out that the reason why my sip2sip account was not being recognized as a trusted caller by the Obi is because I had one of the reserved characters in my username. So in User Settings > User Defined DigitMaps > User Defined DigitMaps2 under DigitMap (if aaaaaaas is my sip2sip username) I had:

(aaaaaaas)

I changed it to:

(aaaaaaa's')

And it went through!!!!!!!!!