Phone display anonymous name and anonymous number

hi. i’m playing with callerid name and number. i got this problem
first of all is maybe something similar to this:
http://forums.asterisk.org/viewtopic.php?p=166952
cause i also ave upgraded 1.6 to 1.8 but i dont know the previous behaviour cause i’m playing with callerid stuff only now… :wink:

i try to explain my problem:
i’m doing some test with callerid functions. i’m doing some call with my mobile phone checking how my xlite client display caller id and number.

if i call whithout che mobile function for hide the number caller id work fine. if i call with the hide number function:

- Executing [<mynumber>@from-pstn:1] NoOp("DAHDI/i1/-6f", "callerid all: "" <>") in new stack -- Executing [<mynumber>@from-pstn:2] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:3] NoOp("DAHDI/i1/-6f", "callerid : 1") in new stack -- Executing [<mynumber>@from-pstn:4] NoOp("DAHDI/i1/-6f", "callerid : iso8859-1") in new stack -- Executing [<mynumber>@from-pstn:5] NoOp("DAHDI/i1/-6f", "callerid : prohib") in new stack -- Executing [<mynumber>@from-pstn:6] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:7] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:8] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:9] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:10] NoOp("DAHDI/i1/-6f", "callerid : "" <>") in new stack -- Executing [<mynumber>@from-pstn:11] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:12] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:13] NoOp("DAHDI/i1/-6f", "callerid : iso8859-1") in new stack -- Executing [<mynumber>@from-pstn:14] NoOp("DAHDI/i1/-6f", "callerid : allowed_not_screened") in new stack -- Executing [<mynumber>@from-pstn:15] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:16] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:17] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:18] NoOp("DAHDI/i1/-6f", "callerid : allowed_not_screened") in new stack -- Executing [<mynumber>@from-pstn:19] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:20] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:21] NoOp("DAHDI/i1/-6f", "callerid : <mynumber>") in new stack -- Executing [<mynumber>@from-pstn:22] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:23] NoOp("DAHDI/i1/-6f", "callerid : ") in new stack -- Executing [<mynumber>@from-pstn:24] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:25] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:26] NoOp("DAHDI/i1/-6f", "callerid : 0") in new stack -- Executing [<mynumber>@from-pstn:27] Set("DAHDI/i1/-6f", "CALLERID(name-pres)=allowed") in new stack -- Executing [<mynumber>@from-pstn:28] Set("DAHDI/i1/-6f", "CALLERID(name)="PIPPO"") in new stack -- Executing [<mynumber>@from-pstn:29] NoOp("DAHDI/i1/-6f", "calleridmod: "PIPPO"") in new stack -- Executing [<mynumber>@from-pstn:30] NoOp("DAHDI/i1/-6f", "calleridmod: allowed") in new stack -- Executing [<mynumber>@from-pstn:31] Dial("DAHDI/i1/-6f", "SIP/2517") in new stack

enabling sip debug on peer 2517 got:

== Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.10.0.101:9332: INVITE sip:2517@10.10.0.101:9332;rinstance=6567dc14deee6245 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.7:5060;branch=z9hG4bK1c80ddec Max-Forwards: 70 [b]From: "Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as251bb8b4[/b] To: <sip:2517@10.10.0.101:9332;rinstance=6567dc14deee6245> Contact: <sip:Anonymous@10.10.0.7:5060> Call-ID: 2101ca3671939c11411afff675b4192a@10.10.0.7:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.6.0 Date: Thu, 01 Mar 2012 15:30:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 228 v=0 o=root 746145371 746145371 IN IP4 10.10.0.7 s=Asterisk PBX 1.8.6.0 c=IN IP4 10.10.0.7 t=0 0 m=audio 19502 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- Called SIP/2517 <--- SIP read from UDP:10.10.0.101:9332 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.0.7:5060;branch=z9hG4bK1c80ddec Contact: <sip:2517@10.10.0.101:9332;rinstance=6567dc14deee6245> To: <sip:2517@10.10.0.101:9332;rinstance=6567dc14deee6245>;tag=2844d545 From: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=as251bb8b4 Call-ID: 2101ca3671939c11411afff675b4192a@10.10.0.7:5060 CSeq: 102 INVITE User-Agent: X-Lite release 1003l stamp 30942 Content-Length: 0

why still
From: "Anonymous"sip:Anonymous@anonymous.invalid
:question: :question: :question:

is a normal behaviour? i can change the callerid for all the kind of external call?

:question: :question: :question:

thanks

think is working doing:
Set(CALLERPRES()=allowed)

now i have just to wonder why…
thanks anyway