asterisk*CLI> sip set debug SIP Debugging enabled asterisk*CLI> asterisk*CLI> asterisk*CLI> <--- SIP read from 88.99.1.202:5060 ---> INVITE sip:201@asterisk.domain.es SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKmmtagrcn Max-Forwards: 70 To: From: "Twinkleeeee" ;tag=eijgk Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 81 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2alpha2 Content-Length: 306 v=0 o=twinkle 680803684 60385702 IN IP4 192.168.1.58 s=- c=IN IP4 192.168.1.58 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (13 headers 14 lines) --- Sending to 88.99.1.202 : 5060 (NAT) Using INVITE request as basis request - jasrmbdqifjzbkh@ibc.domain.lan <--- Reliably Transmitting (NAT) to 88.99.1.202:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKmmtagrcn;received=88.99.1.202;rport=5060 From: "Twinkleeeee" ;tag=eijgk To: ;tag=as54f7f3c5 Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 81 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="578f25f9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'jasrmbdqifjzbkh@ibc.domain.lan' in 32000 ms (Method: INVITE) Found user '200'> asterisk*CLI> <--- SIP read from 88.99.1.202:5060 ---> ACK sip:201@asterisk.domain.es SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKmmtagrcn Max-Forwards: 70 To: ;tag=as54f7f3c5 From: "Twinkleeeee" ;tag=eijgk Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 81 ACK User-Agent: Twinkle/1.2alpha2 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 88.99.1.202:5060 ---> INVITE sip:201@asterisk.domain.es SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKiyhjcsrg Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="578f25f9",uri="sip:201@asterisk.domain.es",response="4df07a0f31e1091bdd3cdba6eed8f9d0",algorithm=MD5 To: From: "Twinkleeeee" ;tag=eijgk Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 82 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2alpha2 Content-Length: 306 v=0 o=twinkle 680803684 60385702 IN IP4 192.168.1.58 s=- c=IN IP4 192.168.1.58 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> --- (14 headers 14 lines) --- Sending to 88.99.1.202 : 5060 (NAT) Using INVITE request as basis request - jasrmbdqifjzbkh@ibc.domain.lan Found user '200' Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.58:8000 Found audio description format speex for ID 98 Found audio description format speex for ID 97 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.58:8000 Looking for 201 in desde-usuarios (domain asterisk.domain.es) list_route: hop: <--- Transmitting (NAT) to 88.99.1.202:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKiyhjcsrg;received=88.99.1.202;rport=5060 From: "Twinkleeeee" ;tag=eijgk To: Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 82 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [201@desde-usuarios:1] Dial("SIP/200-081d8a48", "SIP/201|30|tT") in new stack Really destroying SIP dialog '36af6ca93404016d5fdde7ec0c6e2433@88.99.0.111' Method: INVITE [Jun 19 09:37:52] WARNING[3487]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [201@desde-usuarios:2] Hangup("SIP/200-081d8a48", "") in new stack == Spawn extension (desde-usuarios, 201, 2) exited non-zero on 'SIP/200-081d8a48' Scheduling destruction of SIP dialog 'jasrmbdqifjzbkh@ibc.domain.lan' in 32000 ms (Method: INVITE) asterisk*CLI> <--- Reliably Transmitting (NAT) to 88.99.1.202:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKiyhjcsrg;received=88.99.1.202;rport=5060 From: "Twinkleeeee" ;tag=eijgk To: ;tag=as36b6b21d Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 82 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> asterisk*CLI> <--- SIP read from 88.99.1.202:5060 ---> ACK sip:201@asterisk.domain.es SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKiyhjcsrg Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="578f25f9",uri="sip:201@asterisk.domain.es",response="4df07a0f31e1091bdd3cdba6eed8f9d0",algorithm=MD5 To: ;tag=as36b6b21d From: "Twinkleeeee" ;tag=eijgk Call-ID: jasrmbdqifjzbkh@ibc.domain.lan CSeq: 82 ACK User-Agent: Twinkle/1.2alpha2 Content-Length: 0