SPA-3102 sending bad "From" field

I'm using some SPA-3102's (5.1.7 GW firmware) and am getting "Error 400: BAD REQUEST (multiple values in single-value header) FROM" responses from my PBX.
According to packets grabbed by wireshark, the INVITE from the 3102 goes out with the following FROM field:
From: MACKEY, RON E <sip:[email protected]>;tag=b9c8ff5d2616dc03o1
The ASCII version:
0000   46 72 6f 6d 3a 20 4d 41 43 4b 45 59 2c 20 52 4f From: MACKEY, RO
0010   4e 20 45 20 3c 73 69 70 3a 36 33 30 33 35 37 38 N E <sip:6303578
0020   31 33 35 40 31 39 32 2e 31 36 38 2e 32 2e 31 33 [email protected]
0030   32 3e 3b 74 61 67 3d 62 39 63 38 66 66 35 64 32 2>;tag=b9c8ff5d2
0040   36 31 36 64 63 30 33 6f 31 0d 0a                 616dc03o1..
My reading of the SIP RFC (and it's been a long time since I read an RFC) says that a display-name containing a comma must be enclosed in "".   This would seem to be confirmed by the bad request response from the IP-PBX.
Two questions:
1. Is this an error in the 3102?
2. In either case, is there any way to prevent the 3102 from sending the display-name?
Thanks,
Ron

Thanks for the reply.
Setting "Escape Display Name" to "Yes" had no effect.
In this scenario, the display name is picked up from the incoming phone company Caller ID, so I can't figure out how to blank it out without also losing the caller ID information.
I'm going from a POTS line --> PSTN 1 (on the 3102) --> 3CX (PBX software). I'm using wireshark to verify that the packets coming from the SPA-3102 are just mirroring what's coming in from the phone company. So, in my case, any name that comes in with the form "last name, first name" gets dropped by 3CX as improperly formatted.
Best regards,
Ron

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

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.
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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:
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For the avoidance of doubt, the ATA is always generating the calling *number* field in the CLID spill correctly.
Thanks again!
All the best,
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yytellmey wrote:
Hi,
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Nat mapping enabled=yes
Nat keep alive enabled=yes
Proxy server:correct proxy server specified
Outbound proxy server: Correct outbound proxy server specified
UserID and passwords are configured properly
SIP TAB:
RTP port min=16384
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Thank you for your reply.
Ok, I report my solution because I think that can be userfull for other users.
I have resolved the problem in this manner:
1) on the SPA 3102 I have set the PSTN Line, audio section:
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k) FAX Passthru Method: reinvite;
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d) Echo Canc Enable: NO
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f) Echo Supp Enable: NO
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k) FAX Passthru Method: reinvite
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m) FAX Disable ECAN: NO
n) FAX Enable T38: NO
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The phone attached on the line 1 port of the spa 3102 have not echo when answer or call on the PSTN line as the other phone that have access to PSTN Line.
The phone of the fax macchine have echo, but usualy is not used as phone.
E.

SPA-3102 sending bad "From" field

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