rendered paste body 1. china*CLI> sip debug peer proxy01.sipphone.com 2. SIP Debugging Enabled for IP: 198.65.166.131:5060 3. china*CLI> 4. 5. Sip read: 6. INVITE sip:17476943246:@my.ip.goes.here SIP/2.0 7. Record-Route: <sip:198.65.166.131;r2=on;ftag=84fc4ca7-df13-4968-a804-68dc5dd7f013;lr> 8. Record-Route: <sip:198.65.166.131;transport=tcp;r2=on;ftag=84fc4ca7-df13-4968-a804-68dc5dd7f013;lr> 9. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.ed3635b4.0;i=c85 10. Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 11. Record-Route: <sip:+17759791104@64.211.149.50;lr=True> 12. To: <sip:myaccountgoeshere@sipphone.com> 13. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 14. CSeq: 1 INVITE 15. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 16. Content-Type: application/sdp 17. Content-Length: 254 18. Contact: <sip:+17759791104@64.211.149.50> 19. User-Agent: Callwave SipProxy v2.0 20. Max-Forwards: 16 21. RemoteIP: 64.211.149.50 22. P-hint: usrloc routed 23. 24. v=0 25. o=- 1346543798 1346543798 IN IP4 64.211.149.236 26. s=- 27. t=0 0 28. m=audio 6192 RTP/AVP 18 0 29. c=IN IP4 64.211.149.236 30. a=rtpmap:18 G729/8000 31. a=rtpmap:0 PCMU/8000 32. m=audio 3008 RTP/AVP 96 33. c=IN IP4 64.211.149.50 34. a=rtpmap:96 iLBC/8000 35. a=fmtp:96 mode=30 36. 37. 17 headers, 12 lines 38. Using latest request as basis request 39. Sending to 198.65.166.131 : 5060 (non-NAT) 40. Found peer 'proxy01.sipphone.com' 41. Found RTP audio format 18 42. Found RTP audio format 0 43. Found RTP audio format 96 44. Peer audio RTP is at port 64.211.149.236:3008 45. Found description format G729 46. Found description format PCMU 47. Found description format iLBC 48. Capabilities: us - 0x406 (gsm|ulaw|ilbc), peer - audio=0x504 (ulaw|g729|ilbc)/video=0x0 (nothing), combined - 0x404 (ulaw|ilbc) 49. Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) 50. Looking for 17476943246: in default 51. Reliably Transmitting (no NAT): 52. SIP/2.0 404 Not Found 53. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.ed3635b4.0;i=c85 54. Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 55. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 56. To: <sip:myaccountgoeshere@sipphone.com>;tag=as5b30273a 57. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 58. CSeq: 1 INVITE 59. User-Agent: Asterisk PBX 60. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 61. Contact: <sip:17476943246:@my.ip.goes.here> 62. Content-Length: 0 63. 64. 65. to 198.65.166.131:5060 66. china*CLI> 67. 68. Sip read: 69. ACK sip:17476943246:@my.ip.goes.here SIP/2.0 70. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.ed3635b4.0 71. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 72. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 73. To: <sip:myaccountgoeshere@sipphone.com>;tag=as5b30273a 74. CSeq: 1 ACK 75. Content-Length: 0 76. 77. 78. 7 headers, 0 lines 79. Destroying call '9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0' 80. china*CLI> 81. 82. Sip read: 83. CANCEL sip:17476943246:@my.ip.goes.here SIP/2.0 84. Record-Route: <sip:198.65.166.131;r2=on;ftag=84fc4ca7-df13-4968-a804-68dc5dd7f013;lr> 85. Record-Route: <sip:198.65.166.131;transport=tcp;r2=on;ftag=84fc4ca7-df13-4968-a804-68dc5dd7f013;lr> 86. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 87. Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 88. Record-Route: <sip:+17759791104@64.211.149.50;lr=True> 89. To: <sip:myaccountgoeshere@sipphone.com> 90. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 91. CSeq: 1 CANCEL 92. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 93. Content-Length: 0 94. Max-Forwards: 16 95. P-hint: usrloc routed 96. 97. 98. 13 headers, 0 lines 99. Sending to 198.65.166.131 : 5060 (non-NAT) 100. Transmitting (no NAT): 101. SIP/2.0 481 Call Leg Does Not Exist 102. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 103. Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 104. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 105. To: <sip:myaccountgoeshere@sipphone.com>;tag=as74a51b73 106. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 107. CSeq: 1 CANCEL 108. User-Agent: Asterisk PBX 109. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 110. Contact: 111. Content-Length: 0 112. 113. 114. to 198.65.166.131:5060 115. Destroying call '9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0' 116. china*CLI> 117. 118. 119. Sip read: 120. CANCEL sip:17476943246:@my.ip.goes.here SIP/2.0 121. Record-Route: <sip:198.65.166.131;r2=on;ftag=84fc4ca7-df13-4968-a804-68dc5dd7f013;lr> 122. Record-Route: <sip:198.65.166.131;transport=tcp;r2=on;ftag=84fc4ca7-df13-4968-a804-68dc5dd7f013;lr> 123. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 124. Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 125. Record-Route: <sip:+17759791104@64.211.149.50;lr=True> 126. To: <sip:myaccountgoeshere@sipphone.com> 127. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 128. CSeq: 1 CANCEL 129. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 130. Content-Length: 0 131. Max-Forwards: 16 132. P-hint: usrloc routed 133. 134. 135. 13 headers, 0 lines 136. Sending to 198.65.166.131 : 5060 (non-NAT) 137. Transmitting (no NAT): 138. SIP/2.0 481 Call Leg Does Not Exist 139. Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKfdf3.fd3635b4.0;i=c85 140. Via: SIP/2.0/TCP 64.211.149.50;branch=z9hG4bKfa3b7a8a-a3cc-4a6a-8532-832ec4c337f6 141. From: <sip:+17759791104@192.168.99.204:5062>;tag=84fc4ca7-df13-4968-a804-68dc5dd7f013 142. To: <sip:myaccountgoeshere@sipphone.com>;tag=as7ca411da 143. Call-ID: 9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0 144. CSeq: 1 CANCEL 145. User-Agent: Asterisk PBX 146. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 147. Contact: 148. Content-Length: 0 149. 150. 151. to 198.65.166.131:5060 152. Destroying call '9c921dc4-5b79-45eb-a5c4-78f7124a0da4;cwAttempt=0' 153. china*CLI> 154. china*CLI> 155. 11 headers, 0 lines 156. Reliably Transmitting: 157. OPTIONS sip:198.65.166.131 SIP/2.0 158. Via: SIP/2.0/UDP my.ip.goes.here:5060;branch=z9hG4bK508fd4f4 159. From: "asterisk" <sip:asterisk@my.ip.goes.here>;tag=as7b232ec8 160. To: <sip:198.65.166.131> 161. Contact: <sip:asterisk@my.ip.goes.here> 162. Call-ID: 042a844b6f04aa1e0567e3a37cf98986@my.ip.goes.here 163. CSeq: 102 OPTIONS 164. User-Agent: Asterisk PBX 165. Date: Fri, 25 Aug 2006 17:25:38 GMT 166. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 167. Content-Length: 0 168. 169. (no NAT) to 198.65.166.131:5060 170. china*CLI> 171. 172. Sip read: 173. SIP/2.0 404 Invalid number or user is offline 174. Via: SIP/2.0/UDP my.ip.goes.here:5060;branch=z9hG4bK508fd4f4 175. From: "asterisk" <sip:asterisk@my.ip.goes.here>;tag=as7b232ec8 176. To: <sip:198.65.166.131>;tag=21a483426c2cd5d9b85bffe6bba40a2e.8257 177. Call-ID: 042a844b6f04aa1e0567e3a37cf98986@my.ip.goes.here 178. CSeq: 102 OPTIONS 179. Content-Length: 0 180. 181. 182. 7 headers, 0 lines 183. Destroying call '042a844b6f04aa1e0567e3a37cf98986@my.ip.goes.here' my sip.conf general and peer-specific:[general]context=defaultrealm=my.host.name.hereport=5060bindaddr=one.of.my.hosts.ips.heresrvlookup=yestos=184tos=lowdelayvideosupport=noregister=1747xxxx:pass@proxy01.sipphone.com/1747xxxx <-- also tried no '/1747xxxx' @ end...[proxy01.sipphone.com]type=peerqualify=yeshost=proxy01.sipphone.com