Hello all, We are testing Asterisk 1.8.8.1. In the following scenario, peer 54321 dials 12345:
INVITE sip:12345@10.1.1.88 SIP/2.0 Record-Route: <sip:10.1.1.86;lr=on;ftag=5ebe58983f6c0c84o3> Via: SIP/2.0/UDP 10.1.1.86;branch=z9hG4bKa6c1.4be79d43.0 Via: SIP/2.0/UDP 192.168.4.80:5063;rport=5063;received=10.1.1.86;branch=z9hG4bK-ca013550 From: "54321" <sip:54321@10.1.1.86>;tag=5ebe58983f6c0c84o3 To: <sip:12345@10.1.1.86> Call-ID: adc7c928-b6f6d534@10.1.1.86 CSeq: 102 INVITE Max-Forwards: 69 Contact: "54321" <sip:54321@10.1.1.86:5060> Expires: 240 User-Agent: Linksys/SPA942-6.1.5(a) Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 24058301 24058301 IN IP4 192.168.4.80 s=- c=IN IP4 192.168.4.80 t=0 0 m=audio 16450 RTP/AVP 18 0 2 4 8 96 97 98 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv Peer 54321's caller ID will be blocked, since it enters the following context: [outgoing] exten => _X.,1,Log(NOTICE, Test) exten => _X.,n,Set(CALLERID(num-pres)=prohib_passed_screen) exten => _X.,n,Set(CALLERID(name-pres)=prohib_passed_screen) exten => _X.,n,Dial(SIP/${EXTEN}@Peer) exten => _X.,n,Hangup() When asterisk dials peer 12345, it rewrites the "From" header ("asterisk" <sip:asterisk@10.1.1.88>) instead of keeping it intact. The "Remote-Party-ID" on the other hand, is correct. INVITE sip:12345@10.1.1.87:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.88:5060;branch=z9hG4bK25e11cb8 Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.1.1.88>;tag=as6d2aa852 To: <sip:12345@10.1.1.87:5061> Contact: <sip:asterisk@10.1.1.88:5060> Call-ID: 54deebbf2bb308740e6b9ca817e693a9@10.1.1.88:5060 CSeq: 103 INVITE User-Agent: Asterisk Authorization: Digest username="asterisk", realm="10.1.1.87", algorithm=MD5, uri="sip:12345@10.1.1.87:5061", nonce="6dfc149a8a6801201ba2b28860d6df704f17daeb", response="69d5626fcc5a24980bf641eb1f013813" Date: Thu, 19 Jan 2012 08:57:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "54321" <sip:54321@10.1.1.88>;party=calling;privacy=full;screen=yes Content-Type: application/sdp Content-Length: 525 v=0 o=root 2107042325 2107042326 IN IP4 10.1.1.88 s=m1 c=IN IP4 10.1.1.88 b=CT:384 t=0 0 m=audio 12136 RTP/AVP 18 3 8 0 9 111 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 16724 RTP/AVP 34 98 99 a=rtpmap:34 H263/90000 a=rtpmap:98 h263-1998/90000 a=rtpmap:99 H264/90000 a=sendrecv Please note that in sip.conf we have set: trustrpid = yes sendrpid = yes Any input will be appreciated! Regards, -effie -- _____________________________________________________________________ -- 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