[asterisk-users] SIPAddHeader not working

Jonas Kellens jonas.kellens at telenet.be
Tue Mar 15 04:08:01 CDT 2011


On 03/14/2011 05:06 PM, Steven Howes wrote:
>
> 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

Indeed !

This is the correct INVITE :

/INVITE sip:67121212 at ip_itsp SIP/2.0
Via: SIP/2.0/UDP ip_asterisk:5060;branch=z9hG4bK533e235b;rport
Max-Forwards: 70
From: "32596666" <sip:32596666 at ip_asterisk>;tag=as2c4d672e
To: <sip:67121212 at ip_itsp>
Contact: <sip:32596666 at ip_asterisk>
Call-ID: 60a418e909842287111a1a403498d11b at ip_asterisk
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.16.1
Remote-Party-ID: "32596666" <sip:32596666 at ip_asterisk>;privacy=off;screen=no
Date: Mon, 14 Mar 2011 16:25:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Privacy: id
Content-Type: application/sdp
Content-Length: 266

v=0
o=voip 108024060 108024060 IN IP4 ip_asterisk
s=Asterisk PBX 1.6.2.16.1
c=IN IP4 ip_asterisk
t=0 0
m=audio 11574 RTP/AVP 8 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv/


I notice the presence of the "Privacy: id" SIP header, which is OK !

I also notice the presence of a "Remote-Party-ID" SIPheader... Where 
does this come from ?! Not from my dialplan...



Kind regards,
Jonas.
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