SIP scanners
jazzy:
Quote from: gderf on February 03, 2014, 04:00:18 pm
Why don't you try it without the parentheses and single quotes?
no difference, attached phone still does not ring.
jazzy:
Quote from: drgeoff on February 03, 2014, 04:11:06 pm
Quote from: jazzy on February 03, 2014, 03:24:56 pm
My authusername is Alphanumric.
when I place this in the X inbound call route
{>('myauthusername28'):ph} the attached phone does not ring.
I'm getting myauthusername28 right from SP2 SIP credentials
authusername. Sp2 is using Getonsip (from an IPKALL # )
Does method #4 not work with authusernames from Getonsip?
Your format is correct so the next suspect is that the username you expect is not coming in. Have a look at your call history to see what is shown against an incoming call. Or perhaps call status during an incoming call call throw some light. Revert your InboundCallRoute to ph for those test calls.
Call history in the Obi? When calling in via my cell, the Obi call history shows my CID of the cell phone. No user name. what username might I be looking for?
Maybe this will help. I do not have my GV# ring the Obi.
My GV# forwards to an IPKALL #
The Obi picks up that IPKall number on SP2 using Getonsip credentials.
I currently call out on SP1 ( GV ) at least until May 2014
Testing incoming options for when GV no longer supports XMMP, but want to be sure to
thwart the sip scanners.
drgeoff:
Yes, on reflection I think the Call History only shows "high level" stuff and not far enough down into the nitty-gritty. I'm not even sure if Call Status will have what we are looking for.
Syslog will show the low-level detail but that is a bit more work to set up the server on a PC and then configure the Obi to use it. Also, the log will have many other messages as well as those for an incoming SIP invite.
It is way past my bed-time! I'll be off-line now until morning, UK time.
azrobert:
Here is a trick you can do to determine the username.
Temporarily add {sp1($2)} to the beginning of your SP2 X_InboundCallRoute, then call your SP2 phone number.
$2 is a variable that contains the username of SP2.
The above rule will attempt to bridge the inbound call out SP1 using the username as the outbound phone number.
This call will obviously fail, but the call history will show the username as the outbound number.
jazzy:
@azrobert you da man!
Make the trick described above happen and of course the call failed, but
I did see a 'peer number' show up. It was my alpha numeric user ID, not exactly as the one
in the SIP credentials of SP2 ( it was missing the 'getonsip_' )
Replaced {ph} with {>'my_user_id':ph} and the attached phone now rings! :D
BTW I originally replaced {ph} with {>('my_user_id'):ph} but phone did not ring.
So if any one else uses Getonsip, just delete the 'getonsip_' and put in the rest of your user id.
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