Re: [asterisk-users] SIPAddHeader not working
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)=3259 3259) exten = 67121212,n,SIPAddHeader(P-Preferred-Identity: sip:3259\;user=phone) exten = 67121212,n,SIPAddHeader(Privacy: id) exten = 67121212,n,Dial(SIP/3259/67121212) CLI : INVITE sip:67121...@sip.voip.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-1b5cdb51 From: VC sip:vo...@sip.voip.tld;tag=cb415736707fb109o2 To: sip:67121...@sip.voip.tld Remote-Party-ID: VC sip:vo...@sip.voip.tld;screen=yes;party=calling Call-ID: 2a80707a-bdb7c895@192.168.1.106 CSeq: 101 INVITE Max-Forwards: 70 Contact: VC sip:voip2@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@ip_itsp SIP/2.0 Via: SIP/2.0/UDP ip_asterisk:5060;branch=z9hG4bK533e235b;rport Max-Forwards: 70 From: 3259 sip:3259@ip_asterisk;tag=as2c4d672e To: sip:67121212@ip_itsp Contact: sip:3259@ip_asterisk Call-ID: 60a418e909842287111a1a403498d11b@ip_asterisk CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.16.1 Remote-Party-ID: 3259 sip:3259@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. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a Remote-Party-ID SIPheader... Where does this come from ?! Not from my dialplan... sendrpid in your sip.conf Steve-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
On 03/15/2011 12:24 PM, Steven Howes wrote: On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a Remote-Party-ID SIPheader... Where does this come from ?! Not from my dialplan... sendrpid in your sip.conf Steve Not really : [3259] type=peer host=ip_itsp username=3259 secret=guessthis dtmfmode=rfc2833 canreinvite=no qualify=yes disallow=all allow=alaw allow=gsm amaflags=documentation Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
On 15 Mar 2011, at 11:30, Jonas Kellens wrote: On 03/15/2011 12:24 PM, Steven Howes wrote: On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a Remote-Party-ID SIPheader... Where does this come from ?! Not from my dialplan... sendrpid in your sip.conf Not really : [3259] What about in [general]? 'sip show peer' x might help-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
On 03/15/2011 12:39 PM, Steven Howes wrote: On 15 Mar 2011, at 11:30, Jonas Kellens wrote: On 03/15/2011 12:24 PM, Steven Howes wrote: On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a Remote-Party-ID SIPheader... Where does this come from ?! Not from my dialplan... sendrpid in your sip.conf Not really : [3259] What about in [general]? 'sip show peer' x might help This is the sip show peer on the CLI : vps2301*CLI sip show peer 3259 * Name : 3259 Realtime peer: No Secret : Set MD5Secret: Not set Remote Secret: Not set Context : default Subscr.Cont. : Not set Language : nl AMA flags: DOCUMENTATION Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 2147483647 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID: Yes Subscriptions: Yes Overlap dial : No Forward Loop : Yes DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : ip_itsp Addr-IP : ip_itsp Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 3259 SIP Options : (none) Codecs : 0xa (gsm|alaw) Codec Order : (alaw:20,gsm:20) Auto-Framing : No 100 on REG : No Status : OK (5 ms) Useragent: Reg. Contact : Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs Parkinglot : -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
On 15 Mar 2011, at 15:21, Jonas Kellens wrote: On 03/15/2011 12:39 PM, Steven Howes wrote: On 15 Mar 2011, at 11:30, Jonas Kellens wrote: On 03/15/2011 12:24 PM, Steven Howes wrote: On 15 Mar 2011, at 09:08, Jonas Kellens wrote: I also notice the presence of a Remote-Party-ID SIPheader... Where does this come from ?! Not from my dialplan... sendrpid in your sip.conf Not really : What about in [general]? 'sip show peer' x might help vps2301*CLI sip show peer 3259 snip Send RPID: Yes That's why it's sending RPID. Somewhere in your sip.conf you have sendrpid=yes (wether that's peer specific, or in general). Try setting it to 'no' on the peer? S-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
Hello,
none of the 2 SIP headers are sent...
dialplan :
exten = 67121212,1,NoOp()
exten = 67121212,n,Set(CALLERID(all)=3259 3259)
exten = 67121212,n,SIPAddHeader(P-Preferred-Identity:
sip:3259\;user=phone)
exten = 67121212,n,SIPAddHeader(Privacy: id)
exten = 67121212,n,Dial(SIP/3259/67121212)
CLI :
INVITE sip:67121...@sip.voip.tld SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-1b5cdb51
From: VC sip:vo...@sip.voip.tld;tag=cb415736707fb109o2
To: sip:67121...@sip.voip.tld
Remote-Party-ID: VC sip:vo...@sip.voip.tld;screen=yes;party=calling
Call-ID: 2a80707a-bdb7c895@192.168.1.106
CSeq: 101 INVITE
Max-Forwards: 70
Contact: VC sip:voip2@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
Jonas.
On 03/14/2011 02:48 PM, Andrew Thomas wrote:
Try:
SIPAddHeader(P-Preferred-Identity:sip:${CALLERID(ANI)};user=phone)
SIPAddHeader(Privacy: id)
That works for me in the UK.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 11 March 2011 15:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIPAddHeader not working
Hello,
does anyone have a SIP trace for me where the SIPheader Privacy: id is
present ?? If so, what Asterisk version ?
Kind regards,
Jonas.
On 03/09/2011 06:43 PM, Bryant Zimmerman wrote:
Jonas
In my systems I have seen the Privacy: id when we do our testing but it
has been several months since I have checked it. I am running some tests
later today with one of our customers and I will enable it and do a
capture to confirm but when we do a CID block our vendors say they are
getting the headers correctly
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Jonas Kellensjonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 9:18 AM
To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SIPAddHeader not working
On 03/09/2011 02:09 PM, Bryant Zimmerman wrote:
From: Jonas Kellensjonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
exten = s,n,SIPAddHeader(Privacy: id)
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors pick
it up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2
)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,
1),:,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
I see no great difference. What does
Set(CALLERPRES()=prohib_not_screened) do ?
How does your INVITE look like ? Does the header Privacy: id appears ?
Because it does not in my INVITE.
Kind regards,
Jonas.
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Re: [asterisk-users] SIPAddHeader not working
On 14 Mar 2011, at 15:58, Jonas Kellens wrote: dialplan : exten = 67121212,1,NoOp() exten = 67121212,n,Set(CALLERID(all)=3259 3259) exten = 67121212,n,SIPAddHeader(P-Preferred-Identity: sip:3259\;user=phone) exten = 67121212,n,SIPAddHeader(Privacy: id) exten = 67121212,n,Dial(SIP/3259/67121212) CLI : INVITE sip:67121...@sip.voip.tld SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-1b5cdb51 From: VC sip:vo...@sip.voip.tld;tag=cb415736707fb109o2 To: sip:67121...@sip.voip.tld Remote-Party-ID: VC sip:vo...@sip.voip.tld;screen=yes;party=calling Call-ID: 2a80707a-bdb7c895@192.168.1.106 CSeq: 101 INVITE Max-Forwards: 70 Contact: VC sip:voip2@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-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
Hello,
does anyone have a SIP trace for me where the SIPheader Privacy: id is
present ?? If so, what Asterisk version ?
Kind regards,
Jonas.
On 03/09/2011 06:43 PM, Bryant Zimmerman wrote:
Jonas
In my systems I have seen the Privacy: id when we do our testing but
it has been several months since I have checked it. I am running some
tests later today with one of our customers and I will enable it and
do a capture to confirm but when we do a CID block our vendors say
they are getting the headers correctly
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
*From*: Jonas Kellens jonas.kell...@telenet.be
*Sent*: Wednesday, March 09, 2011 9:18 AM
*To*: brya...@zktech.com, Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] SIPAddHeader not working
On 03/09/2011 02:09 PM, Bryant Zimmerman wrote:
*From*: Jonas Kellens jonas.kell...@telenet.be
*Sent*: Wednesday, March 09, 2011 4:18 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten = s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors
pick it up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
I see no great difference. What does
/Set(CALLERPRES()=prohib_not_screened)/ do ?
How does your INVITE look like ? Does the header /Privacy: id/
appears ? Because it does not in my INVITE.
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] SIPAddHeader not working
Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten = s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0...@sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: VC sip:vo...@sip.domain.be;tag=729476652f511c67o2 To: sip:0...@sip.domain.be Remote-Party-ID: VC sip:vo...@sip.domain.be;screen=yes;party=calling Call-ID: 307124bd-f6881ef@192.168.1.106 CSeq: 101 INVITE Max-Forwards: 70 Contact: VC sip:voip2@192.168.1.106:5063 Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 401 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp/ Using Asterisk 1.6.2.16.1 How do I correctly add the Privacy header ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader not working
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
exten = s,n,SIPAddHeader(Privacy: id)
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors pick it
up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:
,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] SIPAddHeader not working
On 03/09/2011 02:09 PM, Bryant Zimmerman wrote:
*From*: Jonas Kellens jonas.kell...@telenet.be
*Sent*: Wednesday, March 09, 2011 4:18 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Subject*: [asterisk-users] SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten = s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
Using Asterisk 1.6.2.16.1
How do I correctly add the Privacy header ?!
Kind regards,
Jonas.
Jonas
Here is the way we add the rfc-3325 privacey header so our vendors
pick it up correctly. This is what we use in 1.6.x and 1.8.x
When I check on my versions the privacy header appears to be there.
exten = rfc-3325-CPN,1,NoOp(Set Call Privacy)
exten = rfc-3325-CPN,n,NoOp(From ${SIP_HEADER(From)})
exten = rfc-3325-CPN,n,NoOp(To ${SIP_HEADER(To)})
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(SIP_HEADER(From),@,2)})
exten = rfc-3325-CPN,n,GotoIf($[${l_sipheaderfromip} != ]?hasat)
exten =
rfc-3325-CPN,n,Set(l_sipheaderfromip=${CUT(CUT(SIP_HEADER(From),,1),:,2)})
exten = rfc-3325-CPN,n,Goto(gotip)
exten =
rfc-3325-CPN,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),,1),:,1)})
exten = rfc-3325-CPN,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten =
rfc-3325-CPN,n,SIPAddHeader(P-Preferred-Identity:${CALLERID(name)}
sip:+1${CALLERID(num)}@${FROM_IP}\;user=phone)
exten = rfc-3325-CPN,n,SIPAddHeader(Privacy: id)
exten = rfc-3325-CPN,n,Set(CALLERPRES()=prohib_not_screened)
exten = rfc-3325-CPN,n,Set(CALLERID(num)=Anonymous)
exten = rfc-3325-CPN,n,Set(CALLERID(name)=Anonymous)
exten = rfc-3325-CPN,n,Return()
I see no great difference. What does
/Set(CALLERPRES()=prohib_not_screened)/ do ?
How does your INVITE look like ? Does the header /Privacy: id/ appears
? Because it does not in my INVITE.
Kind regards,
Jonas.
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