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P-Asserted Identity / Outbound Caller ID Failure


Alex Kasperavicius
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Hi Guys --

 

Have been playing with your software for a couple of days now and am quite impressed! Very nice job!

 

I am running into a bit of an issue with outbound caller ID to a PSTN SIP trunk provider (flowroute.com).

 

Using Wireshark I discovered that with every outbound call, the P-Asserted Identity is

 

<SIP Username>@<domain>

 

instead of

 

<Account field>@<domain>

 

or even, ideally,

 

ANI@<domain>.

 

My understanding is, and I confirmed with flowroute, is that the P-Asserted identity field should be the 10 or 11 digit ANI (Caller ID)@<domain> unless the privacy bit is set. It is instead being set to my flowroute account number which is problematic.

 

I have been playing with settings for hours and even spoke to the guys at flowroute and they insist that the P-Asserted Identity field is supposed to be the phone number.

 

Is this a bug, are they on crack, or am I missing something? I have been trying to make this work for hours and could really use your help.

 

Thanks much,

 

Alex

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Hi Guys --

 

Have been playing with your software for a couple of days now and am quite impressed! Very nice job!

 

I am running into a bit of an issue with outbound caller ID to a PSTN SIP trunk provider (flowroute.com).

 

Using Wireshark I discovered that with every outbound call, the P-Asserted Identity is

 

<SIP Username>@<domain>

 

instead of

 

<Account field>@<domain>

 

or even, ideally,

 

ANI@<domain>.

 

My understanding is, and I confirmed with flowroute, is that the P-Asserted identity field should be the 10 or 11 digit ANI (Caller ID)@<domain> unless the privacy bit is set. It is instead being set to my flowroute account number which is problematic.

 

I have been playing with settings for hours and even spoke to the guys at flowroute and they insist that the P-Asserted Identity field is supposed to be the phone number.

 

Is this a bug, are they on crack, or am I missing something? I have been trying to make this work for hours and could really use your help.

 

Thanks much,

 

Alex

 

I did not understand completely. But, there are couple of things you can do. One set the "country code" under Domain->Settings. Then under trunk page, you can set the trunk ANI, if you want the same number to go over the trunk all the time. Also, you can play with the "Rewrite global numbers" and "Remote Party/Privacy Indication" fields to see whether you get what you want.

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I did not understand completely. But, there are couple of things you can do. One set the "country code" under Domain->Settings. Then under trunk page, you can set the trunk ANI, if you want the same number to go over the trunk all the time. Also, you can play with the "Rewrite global numbers" and "Remote Party/Privacy Indication" fields to see whether you get what you want.

 

I have tried every possible combination I can think of and in every case the P-Asserted Identity field sent by PBXnSIP is the same thing. It never changes. I really think this is a bug. I should be able to set the identity to my "from" phone number but it's always the username for my SIP account.

 

Should I post clips from the sessions to show what I am describing?

 

 

AK

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Hi :)

 

I agree with Alex. It seems since 3.1.1 some major changes had been implemented. This "country code" + ANI + "global number rewrite" features seems to complicate things :( (just my point of view,as I didnt understand it atm.)

 

But what is very strange to me, it becomes impossible to recreate the "good old" behavior from older versions. An example:

 

If you take a look at the wikipage for OCS the trunk settings for:

 

Remote Party / Privacy Indication: (P-Preferred-Identity)

 

and Assume that call comes from user = primary name of an existing pbxnsip-account (Type = extension) which will be charged for calls from OCS-Mediation-Server to the real world.

 

With this settings it was possible to pass the OCS-Tel-URI via pbxnsip to another trunk for example AudioCodes Mediant 1000 VoiP gateway. So in PSTN your DID arrived at the callee. But now the number from the setting "Assume that call comes from user " is passed for every OCS user or the OCS Conference and Exchange Play on Phone, etc.

 

Since 3.1.1 this seems not to work, or I simply dont understand how to recreate this behavior with the new features...

 

Please advise! Thanks. :)

 

Regards,

Jan

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