-- Executing [s@macro-dial:10] Dial("IAX2/704-1", "SIP/705||tr") in new stack Audio is at 192.168.1.114 port 16078 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.115:5060: INVITE sip:705@192.168.1.115:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK113c6b3f From: "IAX softphone" ;tag=as29787b84 To: Contact: Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 10 Jan 2007 02:33:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 266 v=0 o=root 28323 28323 IN IP4 192.168.1.114 s=session c=IN IP4 192.168.1.114 t=0 0 m=audio 16078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 705 kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> SIP/2.0 100 Trying To: From: "IAX softphone" ;tag=as29787b84 Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK113c6b3f Server: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> SIP/2.0 180 Ringing To: ;tag=de40348691778793i0 From: "IAX softphone" ;tag=as29787b84 Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK113c6b3f Server: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/705-081e7be8 is ringing kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> SIP/2.0 200 OK To: ;tag=de40348691778793i0 From: "IAX softphone" ;tag=as29787b84 Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK113c6b3f Contact: "705" Server: Linksys/SPA941-4.1.15 Content-Length: 210 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 8613858 8613858 IN IP4 192.168.1.115 s=- c=IN IP4 192.168.1.115 t=0 0 m=audio 16400 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (11 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.115:16400 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.115:16400 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.115, port 5060 Transmitting (no NAT) to 192.168.1.115:5060: ACK sip:705@192.168.1.115:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.114:5060;branch=z9hG4bK6579c5c9 From: "IAX softphone" ;tag=as29787b84 To: ;tag=de40348691778793i0 Contact: Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/705-081e7be8 answered IAX2/704-1 kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> INVITE sip:704@192.168.1.114 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-39e14c7d From: ;tag=de40348691778793i0 To: "IAX softphone" ;tag=as29787b84 Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 101 INVITE Max-Forwards: 70 Contact: "705" Expires: 30 User-Agent: Linksys/SPA941-4.1.15 Content-Length: 228 Content-Type: application/sdp v=0 o=- 8614413 8614413 IN IP4 192.168.1.115 s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 16400 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendonly <-------------> --- (12 headers 12 lines) --- Sending to 192.168.1.115 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:16400 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=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 0.0.0.0:16400 Audio is at 192.168.1.114 port 16078 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.1.115:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-39e14c7d;received=192.168.1.115 From: ;tag=de40348691778793i0 To: "IAX softphone" ;tag=as29787b84 Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 28323 28324 IN IP4 192.168.1.114 s=session c=IN IP4 192.168.1.114 t=0 0 m=audio 16078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on IAX2/704-1 -- Stopped music on hold on IAX2/704-1 kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> ACK sip:704@192.168.1.114 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-f780fa35 From: ;tag=de40348691778793i0 To: "IAX softphone" ;tag=as29787b84 Call-ID: 6e54bf4e31c798483d8272ac66298db3@192.168.1.114 CSeq: 101 ACK Max-Forwards: 70 Contact: "705" User-Agent: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> INVITE sip:70@192.168.1.114 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-fcdd3d2e From: "705" ;tag=d22ee91635bde045o0 To: Call-ID: 55ae22a-54c21a41@192.168.1.115 CSeq: 101 INVITE Max-Forwards: 70 Contact: "705" Expires: 240 User-Agent: Linksys/SPA941-4.1.15 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 8614884 8614884 IN IP4 192.168.1.115 s=- c=IN IP4 192.168.1.115 t=0 0 m=audio 16402 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (13 headers 18 lines) --- Sending to 192.168.1.115 : 5060 (no NAT) Using INVITE request as basis request - 55ae22a-54c21a41@192.168.1.115 <--- Reliably Transmitting (no NAT) to 192.168.1.115:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-fcdd3d2e;received=192.168.1.115 From: "705" ;tag=d22ee91635bde045o0 To: ;tag=as72045afe Call-ID: 55ae22a-54c21a41@192.168.1.115 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="35717800" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '55ae22a-54c21a41@192.168.1.115' in 32000 ms (Method: INVITE) Found user '705' kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> ACK sip:70@192.168.1.114 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-fcdd3d2e From: "705" ;tag=d22ee91635bde045o0 To: ;tag=as72045afe Call-ID: 55ae22a-54c21a41@192.168.1.115 CSeq: 101 ACK Max-Forwards: 70 Contact: "705" User-Agent: Linksys/SPA941-4.1.15 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- kiska*CLI> <--- SIP read from 192.168.1.115:5060 ---> INVITE sip:70@192.168.1.114 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-e6ba0696 From: "705" ;tag=d22ee91635bde045o0 To: Call-ID: 55ae22a-54c21a41@192.168.1.115 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="705",realm="asterisk",nonce="35717800",uri="sip:70@192.168.1.114",algorithm=MD5,response="5e998449a2fbd662a88114d490bfad14" Contact: "705" Expires: 240 User-Agent: Linksys/SPA941-4.1.15 Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 8614884 8614884 IN IP4 192.168.1.115 s=- c=IN IP4 192.168.1.115 t=0 0 m=audio 16402 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (14 headers 18 lines) --- Sending to 192.168.1.115 : 5060 (no NAT) Using INVITE request as basis request - 55ae22a-54c21a41@192.168.1.115 Found user '705' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.115:16402 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=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 192.168.1.115:16402 Looking for 70 in from-internal (domain 192.168.1.114) list_route: hop: <--- Transmitting (no NAT) to 192.168.1.115:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.115:5060;branch=z9hG4bK-e6ba0696;received=192.168.1.115 From: "705" ;tag=d22ee91635bde045o0 To: Call-ID: 55ae22a-54c21a41@192.168.1.115 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [70@from-internal:1] Park("SIP/705-08200470", "") in new stack