*CLI> sip show peer 5601 * Name : 5601 Secret : MD5Secret : Context : numberplan-custom-1 Subscr.Cont. : Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : 1 Pickupgroup : 1 Mailbox : 5601 VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : "Alexandre Fiori" <> MaxCallBR : 384 kbps Expire : 1638 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 10.0.5.16 Port 5061 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 5601 SIP Options : (none) Codecs : 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264) Codec Order : (none) Auto-Framing: No Status : Unmonitored Useragent : Ekiga/2.0.11 Reg. Contact : sip:5601@10.0.5.16:5061;transport=udp here's the call debug from sip: <--- SIP read from 10.0.5.16:5072 ---> INVITE sip:6878@10.0.0.7 SIP/2.0 Date: Thu, 31 Jan 2008 13:23:29 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 10.0.5.16:5072;branch=z9hG4bK68192661-6dce-dc11-90e6-0016d393c2e6;rport User-Agent: Ekiga/2.0.11 From: "Alexandre Fiori" ;tag=16762561-6dce-dc11-90e6-0016d393c2e6 Call-ID: 5c712561-6dce-dc11-90e6-0016d393c2e6@kong To: Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 246 Max-Forwards: 70 v=0 o=- 1201785809 1201785809 IN IP4 10.0.5.16 s=Opal SIP Session c=IN IP4 10.0.5.16 t=0 0 m=audio 5036 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 5038 RTP/AVP 31 a=rtpmap:31 H261/90000 <-------------> --- (13 headers 11 lines) --- Sending to 10.0.5.16 : 5072 (NAT) Using INVITE request as basis request - 5c712561-6dce-dc11-90e6-0016d393c2e6@kong Found no matching peer or user for '10.0.5.16:5072' Found RTP audio format 3 Found RTP audio format 101 Found RTP video format 31 Peer audio RTP is at port 10.0.5.16:5036 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Found unknown media description format H261 for ID 31 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.5.16:5036 Looking for 6878 in default (domain 10.0.0.7) <--- Reliably Transmitting (NAT) to 10.0.5.16:5072 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.5.16:5072;branch=z9hG4bK68192661-6dce-dc11-90e6-0016d393c2e6;received=10.0.5.16;rport=5072 From: "Alexandre Fiori" ;tag=16762561-6dce-dc11-90e6-0016d393c2e6 To: ;tag=as24ab869c Call-ID: 5c712561-6dce-dc11-90e6-0016d393c2e6@kong CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Jan 31 11:21:34] NOTICE[20792]: chan_sip.c:13859 handle_request_invite: Call from '' to extension '6878' rejected because extension not found. Scheduling destruction of SIP dialog '5c712561-6dce-dc11-90e6-0016d393c2e6@kong' in 32000 ms (Method: INVITE) <--- SIP read from 10.0.5.16:5072 ---> ACK sip:6878@10.0.0.7 SIP/2.0 CSeq: 1 ACK Via: SIP/2.0/UDP 10.0.5.16:5072;branch=z9hG4bK68192661-6dce-dc11-90e6-0016d393c2e6;rport From: "Alexandre Fiori" ;tag=16762561-6dce-dc11-90e6-0016d393c2e6 Call-ID: 5c712561-6dce-dc11-90e6-0016d393c2e6@kong To: ;tag=as24ab869c Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Length: 0 Max-Forwards: 70 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '5c712561-6dce-dc11-90e6-0016d393c2e6@kong' Method: ACK here's users.conf: [5601] callwaiting = yes context = numberplan-custom-1 fullname = Alexandre Fiori hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes deletevoicemail = no host = dynamic mailbox = 5601 secret = 5601 threewaycalling = yes vmsecret = 5601 registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 disallow = all allow = all here's extensions.conf: ;if I uncomment this and call 123 it works. ;[default] ;exten => 123,1,Playback(busy) [internal] exten => _68XX,1,Dial(h323/${EXTEN}@CCM-CRU,25,wWtT) [numberplan-custom-1] include = internal plancomment = interal parked = no