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Author Topic: Using CSipSimple With OBi  (Read 233167 times)
ianobi
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« on: November 25, 2012, 03:26:42 am »

This thread is only one way of using CSipSimple with OBi. Recent ideas have come up with a different direct calling method with simpler digit maps. See here:
http://www.obitalk.com/forum/index.php?topic=6211.msg39466#msg39466

Anyhow - there's still plenty of interest in this thread - so read on   Smiley

This example uses CSipSimple on an Android smartphone, but similar setups may well be possible with other types of smartphones. CSipSimple is capable of using the native android phone dialler. The aim here is to dial seamlessly from you smartphone through your OBi using only the data connection on your smartphone. It should work exactly like OBiON, except CSipSimple has much better features!

One problem to overcome is that some SIP providers assume that any number starting with 0 should be routed by their servers to PSTN. I got over this by starting all numbers with **7. Also, you need a fixed ip for the OBi or a ddns type address. At the OBi end I used sp2 for incoming calls, my UserAgentPort is 5071.

1. Set up CSipSimple on your cell phone and add a SIP provider as an account (I used sip2sip). You may find it easier to use say an eight digit number as your username.

2. Using CSipSimple > Settings > YourSIPAccount > Filters, set two rules:
   Prefix all numbers with **7.
   Suffix all numbers with @my.ddns.com:5071

3. Set up the InboundCallRoute for the OBi, sp2. I’m using an OBi110, Primary Line PSTN:

Voice Services -> SP2 Service -> X_InboundCallRoute (SP2 must be configured for SIP):

{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**8:>(Mli)),(Mcot)>(<**8:>(Mli)):li},{(Mcot)>(<**7**9:>(Mpp)),(Mcot)>(<**9:>(Mpp)):pp},{(Mcot)>(<**7:>(**0)),(Mcot)>**0:aa},{(Mcot)>(<**7:>(***)),(Mcot)>***:aa2},{(Mcot)>(<**7:>(Mli)),(Mcot)>(Mli):li},{(Mcot)>(<**7:>(0)),(Mcot)>0:ph},{ph}

Mcot has to contain your sip2sip user name.

If you don't have a SIP provider and SP2 is unused, the following will enable SP2 for SIP:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : checked
Voice Services -> SP2 Service -> AuthUserName : (any userid - this is CallerID sent on outgoing calls)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B


I’m sure that you can see the principle – CSipSimple adds **7 to the front of every call, OBi removes **7 from the front of every call and routes the call to where it needs to go. So dialling from your cell phone is the same as dialling from your OBi phone or OBiON.

I also have another sip2sip account set up on my Obi in the sp2 position, but I don’t think this is required or does anything for this set up. I use it for outgoing calls. It does not matter what provider is on sp2, but it must be set up for sip.

The CSipSimple filter rules only work if you choose “integrate with android” and then dial using the cell phone’s dial pad not CsipSimple’s. I guess you could avoid the filter thing if you want to punch in all the extra numbers and letters or store all the numbers in contacts as **7xxxxxxxxxx@my.ddns.com:5071 but the filter rules are there to make life easier!

Testing is best done using a wifi connection.

Everything so far has been about calls coming in from CSipSimple. To call from your OBi to the CSipSimple app set up a speed dial like so:
sp2(userid@sip2sip.info)

For calls coming into say GV on sp1 to ring the phone attached to OBi and your android phone via your sip2sip account on CSipSimple, you need this:

Voice Services -> SP1 Service -> X_InboundCallRoute:
{sp2(userid@sip2sip.info),ph}

Where userid is the sip2sip userid of the account registered with CSipSimple. This will “fork” the incoming call to the android phone and the OBi phone. First device to answer gets the call.


Lots of info there, but I can tell you it does work!

Unrelated: I upgraded to the latest CSipSimple a few days ago and could not make or receive any calls on my lg p500 android oxygen ICS. I downgraded back to 0.04-02r1900 and all works perfectly.

Edit: Recently upgraded to CSipSimple version 1.00.00r2225 and everything works fine.
« Last Edit: June 28, 2013, 07:36:02 am by ianobi » Logged
dinlaca
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« Reply #1 on: November 26, 2012, 08:10:28 pm »

Trying to follow what you did, as it is very similar to what I am trying to do.

I have an android phone that I am trying to setup with my POTS and Google Voice through Obi.  Ideally, the set-up would look something like this tree:

Google Voice (through sp1) --->  Obi110  (through PHONE port) ---> landline plugged into Obi110
                                     (through sp2) ---> to Csipsimple on android phone

In an ideal world, I would be able to dial out of both landline and csipsimple (over data connection), and when someone dialed my Google Voice, it would ring at both landline and android phone.

I tried implementing your setup, and I got (1) sip2sip.info account set up and registered in csipsimple, with appropriate filter programmed (took me a while to find that filter required two steps, "All", then "Add Prefix/Add Suffix"), and (2) Google Voice working fully (call in and call out) on sp1 of my OBi110.  I am having problems programming my user programmed dial plan (the cot in your example) and getting sp2 to work.  Any more explicit details that you can provide after steps 1-2 above would be much appreciated.

Of course, this all assumes that this can be done when sp1 is Google Voice (and not POTS, as yours seems to be set-up).

Any help appreciated.

Thanks.
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ianobi
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« Reply #2 on: November 27, 2012, 08:02:53 am »

Dinlaca,

Welcome to the forum  Smiley

The good news is that what you want can be achieved.  The bad news is that it might take a few days given the problems of us probably being in different time zones and me not being as available as usual over the next few days. Be patient and we will get there. I’ll post enough here to keep you going for a couple of days  Wink

From what you are saying then I’m guessing that your PrimaryLine is set as follows:

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

In your case this is GV. If that is true, then you need to make a small change as follows:

Voice Services -> SP2 Service -> X_InboundCallRoute (SP2 must be configured for SIP):

{(Mcot)>(<**7**1:>(Msp1)),(Mcot)>(<**1:>(Msp1)):sp1},{(Mcot)>(<**7**2:>(Msp2)),(Mcot)>(<**2:>(Msp2)):sp2},{(Mcot)>(<**7**8:>(Mli)),(Mcot)>(<**8:>(Mli)):li},{(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},{ph}

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 your sip2sip user name.

Remember SP2 must be configured for SIP. If you don't have a SIP provider and SP2 is unused, the following will enable SP2 for SIP:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : checked
Voice Services -> SP2 Service -> AuthUserName : (any userid - this is CallerID sent on outgoing calls)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B


Assuming that you are using default settings, then port 5061 will need to be forwarded to the OBi in the router, as this is the default UserAgentPort for sp2. I use 5071 to defeat sip scanners so I forward port 5071. This number must match the suffix in CSipSimple:

@my.ddns.com:5071 is what I have, if you are at default then you should use @my.ddns.com:5061.

It is useful to be familiar with Call History for seeing what is coming into and being sent out of your OBi. This can only be accessed via the web page Status > Call History. The web page IP address can be found by dialling ***1.

I use a softphone (PhonerLite) for testing DigitMaps etc. It is easy to set up accounts with same Caller IDs etc and simulate situations. For instance, in this case you can send in **7**112345678912@ my.ddns.com:5061 and see what happens and look in Call History to see if the call came in on sp2 and went out on sp1.


That’s enough to keep you going for a couple of days  Smiley   I’ll be back Thursday.

Credit goes to RonR for the original explanations, which all of this is based on.

« Last Edit: May 18, 2013, 05:59:21 am by ianobi » Logged
azrobert
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« Reply #3 on: November 27, 2012, 10:23:02 pm »

Ianobi,

Thanks for this tip.  I already had a similar setup using a 2nd ATA, so I was up very quickly.

I setup an account at SIP2SIP using the same username as my ATA and I didn't have to make any changes to my OBi110.

Do you know why the filter doesn't work with the CSipSimple keypad?
I don't like how the native dialer asks if you want to use CSipSimple or Mobile after you dial a number.

Update:

I see the disclaimer under Filters:
"To apply when integration to android is used"

I have an android tablet that doesn't have a dialer, so I can't use CSipSimple with it.  Seems like a stupid restriction. Must be a reason.
« Last Edit: November 28, 2012, 06:40:35 am by azrobert » Logged
ianobi
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« Reply #4 on: November 28, 2012, 08:04:04 am »

Yes, it does seem odd that filters don't work with the CSipSimple keypad. However, using the android native keypad/dialler does mean that you can use all your phone contacts with no need to import to CSipSimple. The downside is an extra choice to make on which service to use - mobile or SIP account.

I still rate CSipSimple quite highly for being very configurable, good choice of codecs etc.
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dinlaca
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« Reply #5 on: November 28, 2012, 01:27:03 pm »

Thank you for your further explanation.

In following your instructions, I am facing a few setbacks.

Setback 1:  Getting my OBi110 to register with sip2sip.info account. 

*********
SOLVED - I followed the settings at the following:  http://www.obitalk.com/forum/index.php?topic=1366.0  I have to remember that search is my FRIEND.  I am keeping below to assist other who may run into analogous problems.
*********

Under System Status --> SP2 Service Status, I am getting the following message:
"Register Failed: 403 This domain is not served here (server=85.17.186.7:5060; retry in 27s)"
The domain 85.17.186.7 resolves to proxy.sipthor.net, from what I can tell.

My related setting changes (I omit the auto changes made in provisioning my Google Voice account) that are different from Default are as follows:

Under ITSP Profile B --> SIP, I have the following:
ProxyServer           proxy.sipthor.net      
ProxyServerPort          (Default - 5060)   
ProxyServerTransport   (Default - UDP)      
RegistrarServer           sip2sip.info
RegistrarServerPort   (Default - 5060)      
UserAgentDomain   sip2sip.info      
OutboundProxy      proxy.sipthor.net   
OutboundProxyPort   (Default - 5060)      
RegistrationPeriod   600

When I tried changing the ProxyServer value to sip2sip.info, I received a similar "Register Failed" error on the System Status --> SP2 Service Status page, but error referenced (server=81.23.228.129)

Under Voice Services --> SP2 Service, the only changes I made from Default are:   

Under submenu "SP2 Service":
      
X_ServProvProfile   B      

X_InboundCallRoute      A "cut and paste" from your DigiMap post above

(Question - Should X_RegisterEnable be checked or unchecked when I am using a real SIP provider (i.e., sip2sip.info)?)

Under submenu "SIP Credentials":

My AuthUserName and AuthPassword are as set up for my sip2sip.info account.

Under User Settings --> User Defined Digit Maps, the only change I made was define a new Digit Map, called "cot" as follows:

(xxxxxxxxxx@sip2sip.info|xxxxxxxxxx) where xxxxxxxxxx is a 10-digit number that is part of the username on my sip2sip.info account.

Setback 2:  Getting my softphone to communicate with my OBi110 box.

This may be related to setback 1 (and probably is), but when I do a call in the format **7(10-digit-number-with-area-code-and-no-leading-1)@(IP-Address-For-My-Router), I get an Address Not Found error or a timeout error.

Anyways, I think that I need to get my OBi to register with my SP2 service (sip2sip.info) before I can move forward.  So, any help that you can provide to me in that regard would be much appreciated.

Please let me know if posting screen shots would be helpful to you in assisting me.

Apologies from a novice.  And, many thanks in advance.
« Last Edit: November 28, 2012, 01:45:41 pm by dinlaca » Logged
QBZappy
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« Reply #6 on: November 28, 2012, 01:46:53 pm »

dinlaca,

Grandstream PBX (Working config)
SIP Server URL    sip2sip.info
Outbound Proxy URL    proxy.sipthor.net
Account Name    2231112222@sip2sip.info
Account ID    2231112222
Authenticate ID 2231112222


dinlaca (Non-working config)
Under ITSP Profile B --> SIP, I have the following:
ProxyServer           proxy.sipthor.net      <--------- You may not need this one
ProxyServerPort          (Default - 5060)   
ProxyServerTransport   (Default - UDP)     
RegistrarServer           sip2sip.info
RegistrarServerPort   (Default - 5060)     
UserAgentDomain   sip2sip.info      <--------- I don't need this one
OutboundProxy      proxy.sipthor.net   
OutboundProxyPort   (Default - 5060)     
RegistrationPeriod   600

You didn't show your log in creditials. Error 403 is a credentials issue I think. Enter them in the form I mentioned above.

See if you can register with this.
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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.
dinlaca
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« Reply #7 on: November 28, 2012, 02:43:05 pm »

dinlaca (Non-working config)
Under ITSP Profile B --> SIP, I have the following:
ProxyServer           proxy.sipthor.net      <--------- You may not need this one
ProxyServerPort          (Default - 5060)   
ProxyServerTransport   (Default - UDP)     
RegistrarServer           sip2sip.info
RegistrarServerPort   (Default - 5060)     
UserAgentDomain   sip2sip.info      <--------- I don't need this one
OutboundProxy      proxy.sipthor.net   
OutboundProxyPort   (Default - 5060)     
RegistrationPeriod   600

Thanks for the further info.

Actually, it turns out that the working login combination is as follows:

Under ITSP Profile B --> SIP, I have the following:
ProxyServer           proxy.sipthor.net
ProxyServerPort          (Default - 5060)   
ProxyServerTransport   (Default - UDP)     
RegistrarServer           proxy.sipthor.net
RegistrarServerPort   (Default - 5060)     
UserAgentDomain   sip2sip.info
OutboundProxy      proxy.sipthor.net   
OutboundProxyPort   (Default - 5060)     
RegistrationPeriod   600

I may or may not need the UserAgentDomain; I haven't tried it without, but since it is working with (per the link I referenced above), I haven't tinkered further.  If I have time later tonight, I will tinker (though I hate to tinker with something working).

Oh, and the specified username does NOT include the "@sip2sip.info" (that is what was causing the 403 errors).

Right now, I have incoming calls to my Google Voice (SP1) forking nicely to my Obi connected phone, and my softphones (both on computer (Telephone for Mac) and on Android (CSipSimple)).  I am still trying to get calls to from my softphones/CSipsimple to go through to my Obi, and I may be making a further post relating to that asking for more info.

Thanks again for all your help. 
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azrobert
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« Reply #8 on: November 28, 2012, 05:55:05 pm »

dinlaca,

This is how I got it working.

CSipSimple --> SIP2SIP -->  Router -->  OBi110  -->  GV or Landline

CSipSimple is sending the call to your router via SIP2SIP.
Set suffix in your phone to "@00.00.00.00:5061"
Where 00.00.00.00 is the external IP address of your router assigned by your ISP.
You can get a dns name assigned to your router, but for now use a hard coded IP address.
If you don't know your IP address go here http://www.whatsmyip.org/

In you router setup Port Range forwarding.
Forward port 5061 to the IP address of your OBi.

Name = anything
Range = 5061 to 5061
Protocol = UDP
IP Address = address

Your OBi IP address should be something like 192.168.1.110

That's it except for the config in your OBi.
You don't need the OBi registered to SIP2SIP.

After you try a call check the OBi call history.

In the left column you should see:
Terminal ID = SP2
Peer Number = Your SIP2SIP UserID.

In the right column you should see:
Terminal ID = GoogleVoice or Line
Peer Number = The number you're calling

If you have one way or no audio port range forward RTP ports 16800 thru 16998 to your OBI same as above.
« Last Edit: November 28, 2012, 06:29:40 pm by azrobert » Logged
dinlaca
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« Reply #9 on: November 28, 2012, 10:13:24 pm »

dinlaca,

This is how I got it working.

CSipSimple --> SIP2SIP -->  Router -->  OBi110  -->  GV or Landline

CSipSimple is sending the call to your router via SIP2SIP.
Set suffix in your phone to "@00.00.00.00:5061"
Where 00.00.00.00 is the external IP address of your router assigned by your ISP.
You can get a dns name assigned to your router, but for now use a hard coded IP address.
If you don't know your IP address go here http://www.whatsmyip.org/

Thanks for the confirmation; this is how I eventually got it working as well.

I want to try and change the hard coded 00.00.00.00 external IP address to dynamic DNS updated domain name (i.e., updated through www.zoneedit.com (where I have grandfathered free dns services), and the Apple AirPort Extreme WAN Bonjour DNS services described here (http://dyn.com/support/airport-time-capsule-with-dynamic-dns/ ), but that is off-topic and will be subject of another thread.

In you router setup Port Range forwarding.
Forward port 5061 to the IP address of your OBi.

Name = anything
Range = 5061 to 5061
Protocol = UDP
IP Address = address

Your OBi IP address should be something like 192.168.1.110

I tried it that way (with 5061).  Then, I opted to do the port forwarding of an alternate port (50xx) to avoid SIP sniffers (per OP).

That's it except for the config in your OBi.
You don't need the OBi registered to SIP2SIP.

Cool that I don't need registration.  But, I figured out the registration issue, and now it is registered (and works for both dial out and dial in).

After you try a call check the OBi call history.

In the left column you should see:
Terminal ID = SP2
Peer Number = Your SIP2SIP UserID.

In the right column you should see:
Terminal ID = GoogleVoice or Line
Peer Number = The number you're calling

If you have one way or no audio port range forward RTP ports 16800 thru 16998 to your OBI same as above.


Call history confirms working.  No one-way audio issues (though it is occasionally a little garbled - best codec to use for this set-up on Csipsimple is . . .?)

Thanks to all of you for such amazing help and direction; I would not have been able to accomplish this without your great help. 

Off to open that next thread about getting DNS services broadcasting, either from OBi (through AirPort Extreme) or from AirPort Extreme.
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dinlaca
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« Reply #10 on: November 29, 2012, 12:50:10 am »

A couple of more observations I have:

If you add a filter of "Directly Call" and "All", then you won't be prompted in native phone dialer to choose between Mobile Phone and SIP account; all calls will be made through the SIP account.

Also, re: the version of Csipsimple which works, I too initially had difficulties getting things to work and stay working with the newer versions.  But, I set up the SIP plan/account with the 1899 version available at nightlies.csipsimple.com/trunk, and then upgraded directly to the 2025 version available there, and have not had a problem since the upgrade (other than sound quality, but I am still working on that codec).

Thanks so much for having motivated me to integrate my cell use (through data plan) into a very functional Google Voice plan through my OBi110.

Now, the last step which would make this PERFECT would be to find a solution that permits both outgoing and incoming calls over the cell phone through cell data plan without use of a third party sip provider (like sip2sip.info); it would have one less step (and one less company to worry about going under), and that much more long term reliability.  But, I will leave that for better minds than me to figure out.
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ianobi
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« Reply #11 on: November 29, 2012, 06:53:01 am »

dinlaca,

Good to see your setup working so quickly. It took me longer than that and I’m still fine tuning!

Your observations on CSipSimple are interesting. I may try an upgrade later using your suggestion.

For further investigations, it is worth looking at your account web page on sip2sip. “History” has a really good SIP debug section to show exactly what happens to a call. When in the debug mode for a call hit the “media” button to show codecs etc.

I’m sure that you have realised that the sip2sip account on your sp2 is there for outbound calls and is not needed for the inbound calls to the OBi. However, it can be used as just another SIP service for any calls coming into sp2.

Quote
Now, the last step which would make this PERFECT would be to find a solution that permits both outgoing and incoming calls over the cell phone through cell data plan without use of a third party sip provider (like sip2sip.info)

I have thought about this, but never managed to make it work! It’s quite easy to set this up using a softphone or ATA where you know the IP addresses. It requires using peer to peer calling without registration. There are examples on this forum. When on home wifi, I can call the android phone at its wifi address using something like sp2(anything@192.168.1.12). I cannot seem to call the other way. This needs more testing – any volunteers?   Smiley
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azrobert
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« Reply #12 on: November 30, 2012, 09:06:19 am »

said by ianobi:
Quote
Now, the last step which would make this PERFECT would be to find a solution that permits both outgoing and incoming calls over the cell phone through cell data plan without use of a third party sip provider (like sip2sip.info)

I have thought about this, but never managed to make it work! It’s quite easy to set this up using a softphone or ATA where you know the IP addresses. It requires using peer to peer calling without registration. There are examples on this forum. When on home wifi, I can call the android phone at its wifi address using something like sp2(anything@192.168.1.12). I cannot seem to call the other way. This needs more testing – any volunteers?
 

All that is needed is a softphone app that doesn't require registration.
The only one I found is Mizudroid, but it doesn't send username with no registration. Don't know if this is a bug or intentional.

Do you know of any softphone apps that don't require registration?   


Update:

My above post isn't true.
You would only have outbound calls from your handset.
Something else would be needed for inbound.
« Last Edit: November 30, 2012, 09:25:50 am by azrobert » Logged
hwittenb
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Posts: 332


« Reply #13 on: November 30, 2012, 08:09:50 pm »

All that is needed is a softphone app that doesn't require registration.
The only one I found is Mizudroid, but it doesn't send username with no registration. Don't know if this is a bug or intentional.

Do you know of any softphone apps that don't require registration?   

The mobile phone softphone app Acrobits (plus their enhanced softphone Groundwire) has an configuration option for a sip account to not register.  In the settings it is a check mark to set the account for outgoing calls only.  The softphone allows you to setup multiple voip accounts so you can have one account like this and a different account that registers for incoming calls. 

This is a report of my Acrobits test today. I setup a direct account on Acrobits to send dialed digits to my OBi110.  I setup the account with my OBi's DynDNS symbolic address and my SP2 sip port as the account proxy server and I set it not to register.  The sip port number is set to forward in my router to the OBi.  This should allow single stage outbound dialing thru the OBi. On SP2 I setup an inbound routing element to send the dialed digits to SP1 based on the incoming caller.  This will bridge the call out thru Google Voice. Calls seemed to work well except for dtmf digits transmitted over the in-progress call after the call was connected.  Maybe the dtmf problem was due to my LG mobile phone.

I have SP2 setup to register to a voip provider.  When I altered the inbound routing on the OBi to try to bridge the call back out thru SP2 or a VGx tied to SP2 there were audio problems and this technique was not satisfactory.  Altering the inbound routing on the OBi to bridge the call on the Line port was OK except for the dtmf problem mentioned above.
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ianobi
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« Reply #14 on: December 01, 2012, 04:52:12 am »

Interesting tests! It would seem that direct peer to peer calling is possible from cell phone to OBi as hwittenb suggests. This does give a free link to the OBi and all its services without a third party being involved. Calls from OBi to a cell phone are always going to need a third party that is registered, as the cell phone IP address will be unknown - except as i suggested if you are at home on home wifi.

Quote
When I altered the inbound routing on the OBi to try to bridge the call back out thru SP2 or a VGx tied to SP2 there were audio problems and this technique was not satisfactory.

I changed this in all my trunks and it seems to solve some problems:

Voice Services > SPX Service > MaxSessions: 4
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hwittenb
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« Reply #15 on: December 01, 2012, 09:55:30 am »

I tried a new Acrobits test today using my OBi202.  I called an unused SP1 setup as RonR outlined with the proxy at 127.0.0.1 with an incoming call routing to bridge the call to SP2 setup as voip.  The bridging worked very well.  The transmission of dtmf after the call completed worked fine better than the OBi110 test.  My guess is the OBi202 has more horsepower than the OBi110.

I also tried calling SP2 and then bridging the call out on SP2.  The call connects and bridges but the RTP sound packets don't start up.  Similiar to what happens with the OBi110.
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ianobi
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Posts: 1828


« Reply #16 on: December 03, 2012, 07:58:39 am »

@ hwittenb.

I can reproduce your findings on my OBi110. If I use a “registered” account on sp2, then all direct calls through trunks and to the OBi phone work fine, but calls bridged through the auto attendant drop out within ten seconds. Unchecking X_RegisterEnable fixes the problem, but then of course you can only make outgoing calls on that service. Using the RonR method seems the best answer:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer : 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID : checked
Voice Services -> SP2 Service -> AuthUserName : (any userid)
Voice Services -> SP2 Service -> X_RegisterEnable : (unchecked)
Voice Services -> SP2 Service -> X_ServProvProfile : B

With this set up we don’t need another sip2sip account on the OBi as outgoing calls from sp2 will go to the android sip2sip account just fine as it is.

It is possible to set up an account in CSipSimple to call without registration. I have done this and made calls both ways with no third parties involved, but only using wifi. I set up a dynamic dns provider on my android phone, but the app tells me I am behind a proxy so it will not work. As things stand we have:

Android phone <> sip2sip <> OBi

The only problems with this setup seems to be that sip2sip does not pass Caller ID and through calls from android phone to OBi need to go out from OBi on a different trunk to the one they come in on. I don't see any dtmf problems.

Plenty to think about  Roll Eyes
« Last Edit: December 03, 2012, 10:44:06 am by ianobi » Logged
QBZappy
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Posts: 2317



« Reply #17 on: December 03, 2012, 08:25:27 am »

Ian,

sip to sip calls should pass CID. That is one of the features of sip.
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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.
ianobi
Hero Member & Beta Tester
*****
Posts: 1828


« Reply #18 on: December 03, 2012, 08:29:26 am »

QBZ,

I agree! Problem is the actual provider sip2sip really does not like us using ui=$1 to pass userid on. I would be happy to be wrong  Smiley
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QBZappy
Hero Member & Beta Tester
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Posts: 2317



« Reply #19 on: December 03, 2012, 08:38:17 am »

Ian,

using ui=$1 to pass userid on.

This code is unique to OBi products. It did not work very well when I tested it with Freephoneline. It showed CID only after 7 rings, making it useless with this SP. I think the CID should be delivered using a more conventional sip method, perhaps using sip uri. I believe that you have tested that without success. Just a stab in the dark, I remember that "X_UseRefer" setting carries the CID in the header in particular use cases. I'm not certain if this is relevant in your case.
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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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