<-- SIP read from 10.0.2.81:5060: ACK sip:%231@10.0.1.185 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKe20afe1514C31388 From: "Christopher Wall" ;tag=CDC3D18F-7A30EC04 To: ;tag=as2ab01f39 CSeq: 1 ACK Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.6.0036 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- dev-asterisk*CLI> <-- SIP read from 10.0.2.81:5060: INVITE sip:%231@10.0.1.185 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKfc959a92F7D7E7CD From: "Christopher Wall" ;tag=CDC3D18F-7A30EC04 To: CSeq: 2 INVITE Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.6.0036 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="1718", realm="asterisk", nonce="572cec2f", uri="sip:%231@10.0.1.185", response="0ad570a8d1af5372e45a9876a3367292", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 243 v=0 o=- 1154451132 1154451132 IN IP4 10.0.2.81 s=Polycom IP Phone c=IN IP4 10.0.2.81 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Using INVITE request as basis request - 8556aab3-c4ab2731-cd77ee56@10.0.2.81 Sending to 10.0.2.81 : 5060 (non-NAT) Found user '1718' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.0.2.81:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for %231 in progrexion (domain 10.0.1.185) Reliably Transmitting (no NAT) to 10.0.2.81:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKfc959a92F7D7E7CD;received=10.0.2.81 From: "Christopher Wall" ;tag=CDC3D18F-7A30EC04 To: ;tag=as2ab01f39 Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Parsing '/etc/asterisk/manager.conf': Found == Manager 'evo_ivr' logged on from 10.0.1.186 == Manager 'evo_ivr' logged off from 10.0.1.186 dev-asterisk*CLI> <-- SIP read from 10.0.2.81:5060: ACK sip:%231@10.0.1.185 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.81;branch=z9hG4bKfc959a92F7D7E7CD From: "Christopher Wall" ;tag=CDC3D18F-7A30EC04 To: ;tag=as2ab01f39 CSeq: 2 ACK Call-ID: 8556aab3-c4ab2731-cd77ee56@10.0.2.81 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.6.0036 Proxy-Authorization: Digest username="1718", realm="asterisk", nonce="572cec2f", uri="sip:%231@10.0.1.185", response="0ad570a8d1af5372e45a9876a3367292", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines)--- Destroying call '8556aab3-c4ab2731-cd77ee56@10.0.2.81' Destroying call 'd2418ed5-2e900773-93ab3248@10.0.2.81' -- Executing AgentLogin("SIP/ev_120-03c0", "") in new stack -- Playing 'agent-user' (language 'en') dev-asterisk*CLI>