コールが失敗した際のログ
extensions.conf を以下のように編集してコールをしてみた。
[default] exten=>2001,1,Dial(SIP/2001,30,r) exten=>2001,2,Hangup()
4444のSIPクライアント側から 2001 に向けてコールしてみる
するとコールが鳴らなかった。
以下はその際に asteriskでデバッグしていたログ
<--- SIP read from UDP:192.168.10.126:3115 ---> <-------------> Really destroying SIP dialog 'YjNmYzZhMDI3MWEzNDY4Njk5ZGVjZThlZGJmZmI0ZGE' Method: ACK <--- SIP read from UDP:192.168.10.126:3115 ---> INVITE sip:2001@192.168.10.127 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.126:3115;branch=z9hG4bK-d8754z-d7df573697413140-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4444@192.168.10.126:3115> To: <sip:2001@192.168.10.127> From: "4444"<sip:4444@192.168.10.127>;tag=aa511b9c Call-ID: NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5 stamp 69607 Content-Length: 249 v=0 o=- 13008754869255619 1 IN IP4 192.168.10.126 s=X-Lite 4 release 4.5 stamp 69607 c=IN IP4 192.168.10.126 t=0 0 m=audio 55584 RTP/AVP 9 8 0 100 101 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 10 lines) --- Sending to 192.168.10.126:3115 (NAT) Using INVITE request as basis request - NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE Found peer '4444' for '4444' from 192.168.10.126:3115 <--- Reliably Transmitting (NAT) to 192.168.10.126:3115 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.126:3115;branch=z9hG4bK-d8754z-d7df573697413140-1---d8754z-;received=192.168.10.126;rport=3115 From: "4444"<sip:4444@192.168.10.127>;tag=aa511b9c To: <sip:2001@192.168.10.127>;tag=as227e2927 Call-ID: NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE CSeq: 1 INVITE Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="47a00bb9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.10.126:3115 ---> ACK sip:2001@192.168.10.127 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.126:3115;branch=z9hG4bK-d8754z-d7df573697413140-1---d8754z-;rport Max-Forwards: 70 To: <sip:2001@192.168.10.127>;tag=as227e2927 From: "4444"<sip:4444@192.168.10.127>;tag=aa511b9c Call-ID: NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:192.168.10.126:3115 ---> INVITE sip:2001@192.168.10.127 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.126:3115;branch=z9hG4bK-d8754z-81bdcf5a6d0141db-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:4444@192.168.10.126:3115> To: <sip:2001@192.168.10.127> From: "4444"<sip:4444@192.168.10.127>;tag=aa511b9c Call-ID: NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite release 4.5 stamp 69607 Authorization: Digest username="4444",realm="asterisk",nonce="47a00bb9",uri="sip:2001@192.168.10.127",response="3cab3c063b2c0eafd34cda726400f858",algorithm=MD5 Content-Length: 249 v=0 o=- 13008754869255619 1 IN IP4 192.168.10.126 s=X-Lite 4 release 4.5 stamp 69607 c=IN IP4 192.168.10.126 t=0 0 m=audio 55584 RTP/AVP 9 8 0 100 101 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (14 headers 10 lines) --- Sending to 192.168.10.126:3115 (NAT) Using INVITE request as basis request - NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE Found peer '4444' for '4444' from 192.168.10.126:3115 == Using SIP RTP CoS mark 5 Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x20000100c (ulaw|alaw|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.10.126:55584 Looking for 2001 in default (domain 192.168.10.127) <--- Reliably Transmitting (NAT) to 192.168.10.126:3115 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.126:3115;branch=z9hG4bK-d8754z-81bdcf5a6d0141db-1---d8754z-;received=192.168.10.126;rport=3115 From: "4444"<sip:4444@192.168.10.127>;tag=aa511b9c To: <sip:2001@192.168.10.127>;tag=as227e2927 Call-ID: NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE CSeq: 2 INVITE Server: Asterisk PBX 1.8.20.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [Mar 26 16:02:16] NOTICE[31239]: chan_sip.c:23272 handle_request_invite: Call from '4444' (192.168.10.126:3115) to extension '2001' rejected because extension not found in context 'default'. Scheduling destruction of SIP dialog 'NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.10.126:3115 ---> ACK sip:2001@192.168.10.127 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.126:3115;branch=z9hG4bK-d8754z-81bdcf5a6d0141db-1---d8754z-;rport Max-Forwards: 70 To: <sip:2001@192.168.10.127>;tag=as227e2927 From: "4444"<sip:4444@192.168.10.127>;tag=aa511b9c Call-ID: NDNkYjc5Y2Y0MmIwMDdkZGFlOTQ4OWVmMjExZTFkZmE CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) ---
asteriskを再起動したら正常にコールされた。
どうやら extensions.conf ファイルを編集したら asterisk を再起動する必要があるらしい。
再起動せずとも読み込ませる方法があるのかもしれないが現段階では分かっていない。（その後 asterisk のコンソールで dialplan reload すればよいことがわかった）
以下はその際に asterisk でデバッグしたログ
== Using SIP RTP CoS mark 5 ←ここから
— Executing [2001@default:1] Dial(“SIP/4444-00000000”, “SIP/2001,30,r”) in new stack
== Using SIP RTP CoS mark 5
— Called SIP/2001
— SIP/2001-00000001 is ringing
— SIP/2001-00000001 answered SIP/4444-00000000
— Locally bridging SIP/4444-00000000 and SIP/2001-00000001
[Mar 26 16:09:08] NOTICE[31534]: res_rtp_asterisk.c:2355 ast_rtp_read: Unknown RTP codec 126 received from ‘192.168.10.126:54574’
[Mar 26 16:09:18] NOTICE[31534]: res_rtp_asterisk.c:2355 ast_rtp_read: Unknown RTP codec 126 received from ‘192.168.10.126:54574’ ←ここまではコールを開始した際に出てきたログ
== Spawn extension (default, 2001, 1) exited non-zero on ‘SIP/4444-00000000’ ←このログは接続を終了した際に出てきた。
