<--- SIP read from 10.1.1.201:5060 ---> INVITE sip:912@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKcc127a5f8f20fa10 From: Max Nemkov ;tag=7a170290 To: 912 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 1 INVITE Contact: Max-Forwards: 70 Subject: SIP Call Content-Type: application/sdp Content-Length: 268 v=0 o=- 423 423 IN IP4 10.1.1.201 s=SIP Session c=IN IP4 10.1.1.201 t=0 0 m=audio 10000 RTP/AVP 8 0 18 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=ptime:60 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (11 headers 13 lines) --- Sending to 10.1.1.201 : 5060 (no NAT) Using INVITE request as basis request - 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 <--- Reliably Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKcc127a5f8f20fa10;received=10.1.1.201 From: Max Nemkov ;tag=7a170290 To: 912 ;tag=as77b85ea7 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="44ecaec7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201' in 32000 ms (Method: INVITE) Found user '914' ch*CLI> <--- SIP read from 10.1.1.201:5060 ---> ACK sip:912@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKcc127a5f8f20fa10 From: Max Nemkov ;tag=7a170290 To: 912 ;tag=as77b85ea7 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- ch*CLI> <--- SIP read from 10.1.1.201:5060 ---> INVITE sip:912@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3 From: Max Nemkov ;tag=7a170290 To: 912 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 2 INVITE Contact: Max-Forwards: 70 Subject: SIP Call Proxy-Authorization: Digest algorithm=MD5,nonce="44ecaec7",realm="asterisk",response="d1b5416c2ee8a222ccd98c65693f9481",uri="sip:912@10.1.1.1:5060",username="914" Content-Type: application/sdp Content-Length: 268 v=0 o=- 423 423 IN IP4 10.1.1.201 s=SIP Session c=IN IP4 10.1.1.201 t=0 0 m=audio 10000 RTP/AVP 8 0 18 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=ptime:60 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (12 headers 13 lines) --- Sending to 10.1.1.201 : 5060 (no NAT) Using INVITE request as basis request - 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 Found user '914' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 10.1.1.201:10000 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.1.1.201:10000 Looking for 912 in internal-phones (domain 10.1.1.1) list_route: hop: <--- Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3;received=10.1.1.201 From: Max Nemkov ;tag=7a170290 To: 912 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [912@internal-phones:1] NoOp("SIP/914-08691000", "") in new stack -- Executing [912@internal-phones:2] Dial("SIP/914-08691000", "SIP/sr/912| 30") in new stack -- Called sr/912 -- SIP/sr-086ae000 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [912@internal-phones:3] Hangup("SIP/914-08691000", "") in new stack == Spawn extension (internal-phones, 912, 3) exited non-zero on 'SIP/914-08691000' Scheduling destruction of SIP dialog '79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3;received=10.1.1.201 rom: Max Nemkov ;tag=7a170290 To: 912 ;tag=as7eef1f27 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> ch*CLI> <--- SIP read from 10.1.1.201:5060 ---> ACK sip:912@10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK7525cb2ed59b3ed3 From: Max Nemkov ;tag=7a170290 To: 912 ;tag=as7eef1f27 Call-ID: 79d9bfda0990b3906c55775f4a9f3c99@10.1.1.201 CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '53e6f47e4dc354155a26c4ec7edee0d6@10.9.0.17' Method: INVITE ch*CLI> <--- SIP read from 10.1.1.201:5060 ---> REGISTER sip:10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKb6071f2c8f04a237 From: 914 ;tag=492253285abc1822134075 To: 914 Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201 CSeq: 15 REGISTER Contact: Max-Forwards: 70 Authorization: Digest algorithm=MD5,nonce="3fd19df6",realm="asterisk",response="45e0553004a8f79d449d555d028d6cdf",uri="sip:10.1.1.1:5060",username="914" Expires: 60 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.1.1.201 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKb6071f2c8f04a237;received=10.1.1.201 From: 914 ;tag=492253285abc1822134075 To: 914 Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201 CSeq: 15 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bKb6071f2c8f04a237;received=10.1.1.201 From: 914 ;tag=492253285abc1822134075 To: 914 ;tag=as4139c298 Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201 CSeq: 15 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6f5f35bc" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5a320266465b32b932cd512702151A1a@10.1.1.201' in 32000 ms (Method: REGISTER) ch*CLI> <--- SIP read from 10.1.1.201:5060 ---> REGISTER sip:10.1.1.1 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK5f863e46180cef41 From: 914 ;tag=492253285abc1822134075 To: 914 Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201 CSeq: 16 REGISTER Contact: Max-Forwards: 70 Authorization: Digest algorithm=MD5,nonce="6f5f35bc",realm="asterisk",response="464986d7f08c27a429bfb23b4b5ebaee",uri="sip:10.1.1.1:5060",username="914" Expires: 60 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.1.1.201 : 5060 (no NAT) <--- Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK5f863e46180cef41;received=10.1.1.201 From: 914 ;tag=492253285abc1822134075 To: 914 Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201 CSeq: 16 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ch*CLI> <--- Transmitting (no NAT) to 10.1.1.201:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.201:5060;branch=z9hG4bK5f863e46180cef41;received=10.1.1.201 From: 914 ;tag=492253285abc1822134075 To: 914 ;tag=as4139c298 Call-ID: 5a320266465b32b932cd512702151A1a@10.1.1.201 CSeq: 16 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 60 Contact: ;expires=60 Date: Fri, 07 Mar 2008 07:13:43 GMT Content-Length: 0 <------------>