Using CSipSimple With OBi

<< < (12/17) > >>

ianobi:
jjjooonnn,

I cannot see any obvious problems with your settings. After a call from CSipSimple to your OBi has finished, what does "Call History" show? You need to look at your local OBi web page (get ip address by dialling ***1. User name and password are both "admin" by default.), then Status > Call History. Does it show the call coming in on SP2? Any "Peer Number" received? If so then we know the routing is ok.

When dialling from your cell phone you should be dialling from the native android dialler, not the CSipSimple Dial pad.

The sip2sip web site has very good call logging information, which might help to see what is being received by their servers and passed on.

If you can, then make the test calls using a wifi connection.

I'm not here much next week, so I hope others will jump in. Keep posting - there is always an answer!

jjjooonnn:
Under Call History it says:
Terminal ID SP2 PHONE1
Peer Number 8134219536
Direction Inbound

When dialing from android, I select a contact (all the numbers are in +18135551234 format) then it asks to use CSip or mobile (pic in link) I use CSip.

I took some more screenshots (call histoy,sip serverlogs and phone): https://plus.google.com/photos/111737898109693132669/albums/5879730141006063857

Thanks again for the help! I'm wondering, maybe it is related to what you said about another sip2sip account? Quote

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.[\quote]

azrobert:
I just looked at your screen shots and you're not prefixing the dialed number with "**1", so I think you need another change.

Is your only requirement routing calls out SP1?

If the answer is yes then change the SP2 Inbound Call Route to:

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

You are comparing your Sip2Sip UserId to Mcot
and
comparing the dialed number to Msp1 plus prefix **7
If you get a match the **7 is stripped off and the call is routed out SP1.
If you don't get a match the call is routed to the Phone Port.

If you are using the default SP1 DigitMap you don't need the **7 prefix.

Remove the **7 prefix in CSipSimple.
You also don't need the User Defined DigitMap cot.

Change the SP2 Inbound Call Route to:
{8134219536>(Msp1):sp1},{ph}

You are comparing your Sip2Sip UserId to 8134219536
and
comparing the dialed number to Msp1.
If you get a match the call is routed out SP1.
If you don't get a match the call is routed to the Phone Port.

jjjooonnn:
@axrobert ;D 8)

Quote

If you are using the default SP1 DigitMap you don't need the **7 prefix.

Remove the **7 prefix in CSipSimple.
You also don't need the User Defined DigitMap cot.

Change the SP2 Inbound Call Route to:
{8134219536>(Msp1):sp1},{ph}

THIS! -Solve it! You're Awesome!

I made the changes above and everything chimed together!

My phone set up is now, as follows (for your consideration):
I'm using a $100 tmobile 1,000 minutes card on a smart phone with the calls getting forwarded by google voice; the only down side is that I don't have data for google maps or pandora when driving, it's only for calls (texts aren't forwarded).
Then I can use this on college/home wifi all from the same phone,
And my OBi line to use for those important calls at the house.
This will save me around $380 a year in cell phone bills, I'm hoping everyone can see how cool this is!

Here's some newbie follow up question (feel free to ignore, I'm very grateful for you guys' help):

What's best for fast/slow connections, SILK 24/G729?

And do these codecs need to be compatible with the OBi's codecs or google voice's servers?

ianobi, so from what I read it looks like STUN/ICE is a no-go huh?

Navigation

[0] Message Index

[#] Next page

[*] Previous page