asterisk*CLI> Sip read: INVITE sip:2166192000@192.168.1.5;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7a900c6b778C8F4E From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: CSeq: 1 INVITE Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.3.4 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1105772802 1105772802 IN IP4 192.168.1.103 s=Polycom IP Phone c=IN IP4 192.168.1.103 t=0 0 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 14 headers, 10 lines Using latest request as basis request Sending to 192.168.1.103 : 5060 (non-NAT) 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 192.168.1.103:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ula w|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7a900c6b778C8F4E From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as7003d278 Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="0260d22d" Content-Length: 0 to 192.168.1.103:5060 Scheduling destruction of call '19b593a1-e187fc97-49f09464@192.168.1.103' in 15000 ms Found user '100' asterisk*CLI> Sip read: ACK sip:2166192000@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7a900c6b778C8F4E From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as7003d278 CSeq: 1 ACK Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.3.4 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines asterisk*CLI> Sip read: INVITE sip:2166192000@192.168.1.5;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7aa9dfd0D386D3C3 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: CSeq: 2 INVITE Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.3.4 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="100", realm="asterisk", nonce="0260d22d", uri="sip:2166192000@192.168.1.5;user=phone", response="41c7d41d29c34048dcea1f4cbf4e31f9", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1105772802 1105772802 IN IP4 192.168.1.103 s=Polycom IP Phone c=IN IP4 192.168.1.103 t=0 0 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, 10 lines Using latest request as basis request Sending to 192.168.1.103 : 5060 (non-NAT) 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 192.168.1.103:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ula w|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Found user '100' Looking for 2166192000 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7aa9dfd0D386D3C3 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as20db1a7d Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.103:5060 -- Executing ChanIsAvail("SIP/100-d8fb", "Zap/1") in new stack -- Hungup 'Zap/1-1' -- Executing Dial("SIP/100-d8fb", "IAX2/boehnlein@pbx1/2166192000") in new stack -- Called boehnlein@pbx1/2166192000 -- Call accepted by 207.166.192.188 (format ulaw) -- Format for call is ulaw We're at 192.168.1.5 port 19968 Answering with preferred capability 0x4 (ulaw) Transmitting (no NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7aa9dfd0D386D3C3 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as20db1a7d Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 158 v=0 o=root 20888 20888 IN IP4 192.168.1.5 s=session c=IN IP4 192.168.1.5 t=0 0 m=audio 19968 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - to 192.168.1.103:5060 -- IAX2/pbx1/5 is ringing 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:103@192.168.1.118:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK7db5c288 From: "2169203021" ;tag=as03cca149 To: Contact: Call-ID: 19aa33402f4a308c288d28c152a5d624@192.168.1.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sat, 15 Jan 2005 07:06:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.1.118:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:102@192.168.1.118:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK0ac6c9e1 From: "2169203021" ;tag=as50bdacab To: Contact: Call-ID: 1c5369ba11c7a7be417fcce774ff264b@192.168.1.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sat, 15 Jan 2005 07:06:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.1.118:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK To: ;tag=6f1a7a7be90f07cbi1 From: "2169203021" ;tag=as03cca149 Call-ID: 19aa33402f4a308c288d28c152a5d624@192.168.1.5 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK7db5c288 Server: Sipura/SPA2000-2.0.10(e) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 10 headers, 0 lines Destroying call '19aa33402f4a308c288d28c152a5d624@192.168.1.5' ip read: CLI> SIP/2.0 200 OK To: ;tag=eba877fbb8dd284bi0 From: "2169203021" ;tag=as50bdacab Call-ID: 1c5369ba11c7a7be417fcce774ff264b@192.168.1.5 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK0ac6c9e1 Server: Sipura/SPA2000-2.0.10(e) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 10 headers, 0 lines Destroying call '1c5369ba11c7a7be417fcce774ff264b@192.168.1.5' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:101@192.168.1.114:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK26138c6e From: "2169203021" ;tag=as5934e5f2 To: Contact: Call-ID: 566b387b036c2e7b1c3dc02d70156bbb@192.168.1.5 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Sat, 15 Jan 2005 07:06:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.1.114:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.5:5060;branch=z9hG4bK26138c6e From: "2169203021" ;tag=as5934e5f2 To: ;tag=00078535613f0f43239fddd5-48c9b0db Call-ID: 566b387b036c2e7b1c3dc02d70156bbb@192.168.1.5 Date: Sat, 15 Jan 2005 07:07:02 GMT CSeq: 102 OPTIONS Server: CSCO/7 Content-Type: application/sdp Content-Length: 237 Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER v=0 o=Cisco-SIPUA 0 0 IN IP4 192.168.1.114 s=SIP Call c=IN IP4 192.168.1.114 t=0 0 m=audio 1 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 a=fmtp:101 0-15 11 headers, 11 lines Destroying call '566b387b036c2e7b1c3dc02d70156bbb@192.168.1.5' -- IAX2/pbx1/5 stopped sounds -- IAX2/pbx1/5 answered SIP/100-d8fb We're at 192.168.1.5 port 19968 Answering with preferred capability 0x4 (ulaw) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK7aa9dfd0D386D3C3 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as20db1a7d Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 158 v=0 o=root 20888 20889 IN IP4 192.168.1.5 s=session c=IN IP4 192.168.1.5 t=0 0 m=audio 19968 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - to 192.168.1.103:5060 asterisk*CLI> Sip read: ACK sip:2166192000@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK4d4951efB8C0DC12 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as20db1a7d CSeq: 2 ACK Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.3.4 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines asterisk*CLI> Sip read: BYE sip:2166192000@192.168.1.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK40108ba56C7F1BA8 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as20db1a7d CSeq: 3 BYE Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.3.4 Proxy-Authorization: Digest username="100", realm="asterisk", nonce="0260d22d", uri="sip:2166192000@192.168.1.5;user=phone", response="41c7d41d29c34048dcea1f4cbf4e31f9", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Sending to 192.168.1.103 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK40108ba56C7F1BA8 From: "Gregory Boehnlein" ;tag=57505F4D-22FE273A To: ;tag=as20db1a7d Call-ID: 19b593a1-e187fc97-49f09464@192.168.1.103 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.1.103:5060 -- Hungup 'IAX2/pbx1/5' == Spawn extension (default, 2166192000, 2) exited non-zero on 'SIP/100-d8fb' Destroying call '19b593a1-e187fc97-49f09464@192.168.1.103'