[asterisk-users] SIPAddHeader not working

[Steven Howes]

On 14 Mar 2011, at 15:58, Jonas Kellens wrote:
> dialplan :
> 
> exten => 67121212,1,NoOp()
> exten => 67121212,n,Set(CALLERID(all)="32596666" <32596666>)
> exten => 67121212,n,SIPAddHeader(P-Preferred-Identity: <sip:32596666\;user=phone>)
> exten => 67121212,n,SIPAddHeader(Privacy: id)
> exten => 67121212,n,Dial(SIP/32596666/67121212)
> 
> 
> CLI :
> 
> INVITE sip:67121212 at sip.voip.tld SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-1b5cdb51
> From: "VC" <sip:voip2 at sip.voip.tld>;tag=cb415736707fb109o2
> To: <sip:67121212 at sip.voip.tld>
> Remote-Party-ID: "VC" <sip:voip2 at sip.voip.tld>;screen=yes;party=calling
> Call-ID: 2a80707a-bdb7c895 at 192.168.1.106
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: "VC" <sip:voip2 at 192.168.1.106:5063>
> Expires: 240
> User-Agent: Linksys/SPA941-5.1.8
> Content-Length: 399
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: replaces
> Content-Type: application/sdp

That's the invite from the phone, not from Asterisk... no?

S
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