Disable calling name presentation on SPA-3102

[methorpe24]

Hi,

If I send a SIP INVITE to my SPA-3102, where the From header is like this -- (spaces inserted to stop the forum software treating it as an email address -- they're not there in the real invite)

From: Caller Name <01234567890 @ my.sip.server.net>;tag=as4b617ab1

-- the SPA-3102 generates a Caller ID spill on its FXS port with 'Caller Name' as the calling name, and '01234567890' as the calling number. That's all well and good.

If the From: header doesn't have a caller name, but is like this instead --

From: <01234567890 @ my.sip.server.net>;tag=as4b617ab1

-- the box sets the calling name to be 01234567890 as well.

Is there any way to turn that off, and have the SPA just not present a calling name at all?

If not, no bother! I'm just trying to get my box to behave a little more like BT with regards to caller ID presentation -- they don't ever send a reason for no calling *name*, but if the calling number is withheld or unavailable they will set the calling name to Withheld or Unavailable -- and set a reason for no calling number.

Many thanks!

Martin

Message was edited by: Martin Thorpe -- hopefully removed the auto-'email address' tagging! (Argh, no, it didn't. Bodged a different way.)

[lifennel]

Martin,

Thanks for the thoughtful presentation of your SIP headers, that definitely helps me in my lab setting. I see the same thing too, when I'm using an Asterisk server. I believe it is the expected behavior of this ATA to handle a FROM field with no string of text identifying caller ID, to just display the string of numbers in the sip URI. What I'm interested in, is, what does your entire SIP packet look like? Are we using a Contact field?

But, again, I believe it is expected behavior and I do not believe there is a setting to allow the ATA that, in the event of no string of text being available in the FROM field, to "failover" to display "Unknown" or something like that, unless the product management team has any ideas.

Cheers,

Lindsey

[methorpe24]

Hi Lindsey,

Thanks for the quick response. Here's a complete SIP invite -- I've changed the telephone number and put spaces around @ signs again, but everything else is unmodified.

INVITE sip:spa-line1 @ 81.2.113.115:5060 SIP/2.0

Via: SIP/2.0/UDP 81.187.239.177:5060;branch=z9hG4bK4062e0e9;rport

Max-Forwards: 70

From: <01234567890>;tag=as75e22314

To:

Contact: <01234567890>

Call-ID: 445f75c33908fff74829a514159e9946 @ sentry.met24.net

CSeq: 102 INVITE

User-Agent: Asterisk

Date: Mon, 29 Oct 2012 19:51:07 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 286

So there is a contact field in there as well.

That's from a slightly patched Asterisk server, which doesn't put a calling name in if it's blank -- by default if you didn't set a calling name, Asterisk will also set the calling name from the calling number and you'd get this instead:

From: "01234567890" <01234567890>;tag=as54c7bb08

I've done product management myself so I know one customer asking for it to work a little differently (as opposed to it doing something wrong!) isn't going to make a change -- that's no problem at all. If it were to be changed, I'd rather the ATA didn't generate a calling name field in the CLID spill at all, rather than 'Unknown'. But hey, that's just my opinion!

For the avoidance of doubt, the ATA is always generating the calling *number* field in the CLID spill correctly.

Thanks again!

All the best,

Martin