Using CSipSimple With OBi
dinlaca:
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.
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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.
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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.
QBZappy:
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.
dinlaca:
Quote from: QBZappy on November 28, 2012, 01:46:53 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.
azrobert:
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.
dinlaca:
Quote from: azrobert 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/
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.
Quote from: azrobert on November 28, 2012, 05:55:05 pm
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).
Quote from: azrobert on November 28, 2012, 05:55:05 pm
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).
Quote from: azrobert on November 28, 2012, 05:55:05 pm
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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