In extensions.conf: exten => 731,1,Set(CALLERID(num)=123) exten => 731,n,Dial(SIP/123@66.212.134.192,60,) Asterisk 1.2.18 Same happens in 1.4.4 -- Starting simple switch on 'Zap/3-1' -- Executing Set("Zap/3-1", "CALLERID(num)=123") in new stack -- Executing Dial("Zap/3-1", "SIP/123@66.212.134.192|60|D(22622#1112223334#)") in new stack We're at 192.168.67.18 port 11210 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 66.212.134.192:5060: INVITE sip:123@66.212.134.192 SIP/2.0 Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK1a7d2f9c;rport From: "Channel 3" ;tag=as7b39bbfa To: Contact: Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 22 Jun 2007 17:53:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 265 v=0 o=root 14976 14976 IN IP4 192.168.67.18 s=session c=IN IP4 192.168.67.18 t=0 0 m=audio 11210 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 123@66.212.134.192 pungenday*CLI> <-- SIP read from 66.212.134.192:5060: SIP/2.0 180 Ringing From: "Channel 3";tag=as7b39bbfa To: ;tag=a000217-13c4-467bc56e-13c8fe99-587d9c53 Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK1a7d2f9c Contact: Content-Length: 0 --- (8 headers 0 lines) --- -- SIP/66.212.134.192-081b5458 is ringing pungenday*CLI> <-- SIP read from 66.212.134.192:5060: SIP/2.0 403 Forbidden From: "Channel 3";tag=as7b39bbfa To: ;tag=a000217-13c4-467bc56e-13c8fe99-587d9c53 Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.67.18:5060;received=212.179.75.202;rport=5060;branch=z9hG4bK1a7d2f9c Content-Length: 0 --- (7 headers 0 lines) --- Transmitting (no NAT) to 66.212.134.192:5060: ACK sip:123@66.212.134.192 SIP/2.0 Via: SIP/2.0/UDP 192.168.67.18:5060;branch=z9hG4bK1a7d2f9c;rport From: "Channel 3" ;tag=as7b39bbfa To: ;tag=a000217-13c4-467bc56e-13c8fe99-587d9c53 Contact: Call-ID: 39962e1c067c58876c5af1240a19b112@192.168.67.18 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jun 22 20:53:19 WARNING[14990]: chan_sip.c:9847 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Channel 3" ;tag=as7b39bbfa' -- SIP/66.212.134.192-081b5458 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'Zap/3-1' status is 'CONGESTION' Destroying call '39962e1c067c58876c5af1240a19b112@192.168.67.18' pungenday*CLI> sip no debug SIP Debugging Disabled -- Hungup 'Zap/3-1'