<--- Transmitting (no NAT) to 192.168.2.237:8160 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;received=192.168.2.237;rport=8160 From: "Phillip 45";tag=394d9618 To: Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [2299614@numberplan-custom-1:1] Macro("SIP/46-09022000", "trunkdial|Zap/g1/2299614") in new stack -- Executing [s@macro-trunkdial:1] Dial("SIP/46-09022000", "Zap/g1/2299614") in new stack [May 30 18:57:22] WARNING[34627]: channel.c:3084 ast_request: No translator path exists for channel type Zap (native 76) to 256 [May 30 18:57:22] WARNING[34627]: app_dial.c:1099 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-trunkdial:2] Goto("SIP/46-09022000", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-trunkdial,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp("SIP/46-09022000", "No pude relaizar la llamada: CHANUNAVAIL| 0") in new stack == Auto fallthrough, channel 'SIP/46-09022000' status is 'CHANUNAVAIL' <--- Transmitting (no NAT) to 192.168.2.237:8160 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;received=192.168.2.237;rport=8160 From: "Phillip 45";tag=394d9618 To: ;tag=as024ce994 Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> pbx*CLI> <--- SIP read from 192.168.2.237:8160 ---> INVITE sip:2299614@192.168.2.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;rport Max-Forwards: 70 Contact: To: From: "Phillip 45";tag=394d9618 Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="46",realm="asterisk",nonce="29d290c4",uri="sip:2299614@192.168.2.6",response="d0944b2b218082dd5b5f37cc4c0142d9",algorithm=MD5 Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 535 v=0 o=- 26720725 26720742 IN IP4 192.168.2.237 s=eyeBeam c=IN IP4 192.168.2.237 t=0 0 m=audio 8144 RTP/AVP 18 3 0 100 6 8 99 98 97 5 102 101 a=alt:1 1 : E681ABC9 0000007D 192.168.2.237 8144 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:99 evrc/8000 a=rtpmap:98 ilbc/8000 a=rtpmap:97 speex/8000 a=rtpmap:102 l16/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv m=video 41934 RTP/AVP 105 34 a=alt:1 1 : 98EEFBCB 00000002 192.168.2.237 41934 a=fmtp:105 profile=3;level=20 a=rtpmap:105 h263-2000/90000 a=sendrecv <-------------> --- (14 headers 20 lines) --- Ignoring this INVITE request [May 30 18:57:22] NOTICE[34627]: chan_sip.c:13671 handle_request_invite: Unable to create/find SIP channel for this INVITE <--- Transmitting (no NAT) to 192.168.2.237:8160 ---> SIP/2.0 503 Unavailable Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;received=192.168.2.237;rport=8160 From: "Phillip 45";tag=394d9618 To: ;tag=as024ce994 Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'a9646d79260ae028@VG9zaGliYVBuZXVtYW5u' in 32000 ms (Method: INVITE) Really destroying SIP dialog 'a9646d79260ae028@VG9zaGliYVBuZXVtYW5u' Method: INVITE pbx*CLI> <--- SIP read from 192.168.2.237:8160 ---> ACK sip:2299614@192.168.2.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;rport To: ;tag=as024ce994 From: "Phillip 45";tag=394d9618 Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from 192.168.2.237:8160 ---> ACK sip:2299614@192.168.2.6 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.237:8160;branch=z9hG4bK-d87543-994d885ee40d8e4c-1--d87543-;rport To: ;tag=as024ce994 From: "Phillip 45";tag=394d9618 Call-ID: a9646d79260ae028@VG9zaGliYVBuZXVtYW5u CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- pbx*CLI> <--- SIP read from 192.168.2.237:8160 --->