new Caller-ID spoofing inquiry over SIP for Obi 110
hwittenb:
rob613,
The X_UserAgentName field is to tell the distant server about your OBi adapter ... for instantance that it is an OBi110 running firmware 1.3.0.2744.
On a straight outbound call, in the Sip Invite the VoiceGateway (if using) AuthID goes into the From: and the Remote-Party-ID: fields. The SPx AuthUserName goes into the Contact field. This is also true if ringing a distant number from the Inbound Routing.
On the other hand, if you are forwarding a call to a Sip Uri using SPx (and have X_SpoofCallerID enabled) the CallerID goes into the From: and Remote-Party-ID: fields, the SPx AuthUserName goes into the Contact field, assuming the OBi captured the incoming caller ID. If you are forwarding a call to a Sip Uri using VGx the OBi does not forward the call, you need to specify SPx.
The toll free call example mentioned early on in the thread works because the distant server (tf.callwithus.com) does not authenticate the call (determine that you do have an account with them) and the AuthID you setup in the VGx goes into the From: and the Remote-Party-ID fields.
If you want to study how different variables affect the sip signalling you should setup to use the packet sniffer WireShark. You can look at the packets that actually come out of the adapter. You setup your OBi adapter and your computer to both attach to an old fashioned hub (not a switch) that is cabled to your external router. With that cabling your computer can see all the ethernet traffic going to and from your OBi and the internet. Then you install WireShark (a free program) on your computer. WireShark has a nice formatting feature for sip calls. After you make a call you can invoke the WireShark formatter which will outline the call signalling and then you can click on a specific packet (like the sip INVITE) and see the various fields.
tmetro:
Quote from: QBZappy on November 21, 2012, 03:04:39 pm
Quote from: tmetro on November 21, 2012, 02:21:26 pm
...the expression becomes:
{(Mtesting):vg3($1>9991234)}
$1 is used internally (temp incoming CID variable) for the callback feature. ...I don't think it can be used in any user configuration call strategy.
So would you agree that if it doesn't work as depicted above, the firmware or the documentation has a bug?
Does Obihai have a bug reporting mechanism, or do we just hope they notice mention of bugs in forum threads?
-Tom
tmetro:
Quote from: rob613 on November 22, 2012, 01:13:17 am
I also discovered that although I am using a VG (which is based on my SP2 configuration) I am seeing my SP2 username, which should be irrelevant, getting exposed to the forwarded SIP call.
Sounds like a bug.
Quote from: hwittenb on November 22, 2012, 10:54:23 am
If you are forwarding a call to a Sip Uri using VGx the OBi does not forward the call, you need to specify SPx.
Sounds like a bug.
Quote
If you want to study how different variables affect the sip signalling you should setup to use the packet sniffer WireShark.
Good general advice. In this case Rob is using a developer account at Tropo (free) to receive the SIP invite messages, and it logs the full headers for easy inspection.
-Tom
QBZappy:
rob613,
This might be the solution to pass CID. Have a look.
ui=$1
Re: Spoofing Solved
http://www.obitalk.com/forum/index.php?topic=4658.msg30451#msg30451
Quote from: azrobert on November 27, 2012, 12:25:20 pm
I never could get spoofing to work. I gave it one more try and I did it!
This works WITHOUT enabling X_SpoofCallerID. The problem with X_SpoofCallerID is that some providers don't like spoofing and when enabled it's on for everything going out SPx.
I got it to work with PBX in a Flash and Callcentric.
Here's an InboundCallRoute example:
{ph,sp2(100@192.168.1.999:5060;ui=$1),sp2(17772223333@in.callcentric.com;ui=$1}
rob613:
QBZappy, thank you very much for following up with this information.
It did not work for me.
Putting the ";ui=$1" expression within the call through the VG (where it only takes an extension, not a full SIP specification) resulted in it getting inserted as part of the extension, literally. I also tried putting it elsewhere in the VG call.
For my need I would like to see something like an even more generic SIP forward option where, similar to a VG, no registration is used, but where all options for a SIP call could be specified, including ui, and spoofing.
This would also help reduce the exposure of the SP2 username when calling out through a VG that has nothing in common with the SP2 configuration except that it is the same SIP protocol - I see this as a bug.
FYI, I have solved my problem a different way - a bit complicated but it seems to work. Anyone who wants to know more can try to get a private message to me. It requires external help, and runs not just within the Obi.
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