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  1. <--- SIP read from UDP:192.168.10.229:5060 --->
  2. INVITE sip:[email protected] SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ece9950c31089f;rport
  4. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  5. To: <sip:[email protected]>
  6. Call-ID: [email protected]
  7. CSeq: 189 INVITE
  8. Contact: <sip:[email protected]:5060>
  9. Content-Type: application/sdp
  10. Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
  11. Max-Forwards: 70
  12. Supported: 100rel, replaces, from-change
  13. User-Agent: SIPPER for PhonerLite
  14. P-Preferred-Identity: <sip:[email protected]>
  15. Content-Length: 447
  16.  
  17. v=0
  18. o=- 1929311701 1 IN IP4 192.168.10.229
  19. s=SIPPER for PhonerLite
  20. c=IN IP4 192.168.10.229
  21. t=0 0
  22. m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 101
  23. a=rtpmap:107 opus/48000
  24. a=rtpmap:8 PCMA/8000
  25. a=rtpmap:0 PCMU/8000
  26. a=rtpmap:2 G726-32/8000
  27. a=rtpmap:3 GSM/8000
  28. a=rtpmap:97 iLBC/8000
  29. a=rtpmap:110 speex/8000
  30. a=rtpmap:111 speex/16000
  31. a=rtpmap:9 G722/8000
  32. a=rtpmap:101 telephone-event/8000
  33. a=fmtp:101 0-16
  34. a=ssrc:2188520094
  35. a=sendrecv
  36. <------------->
  37. --- (14 headers 19 lines) ---
  38. Sending to 192.168.10.229:5060 (no NAT)
  39. Sending to 192.168.10.229:5060 (no NAT)
  40. Using INVITE request as basis request - [email protected]
  41. Found peer '413' for '413' from 192.168.10.229:5060
  42.  
  43. <--- Reliably Transmitting (no NAT) to 192.168.10.229:5060 --->
  44. SIP/2.0 401 Unauthorized
  45. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ece9950c31089f;received=192.168.10.229;rport=5060
  46. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  47. To: <sip:[email protected]>;tag=as440c3211
  48. Call-ID: [email protected]
  49. CSeq: 189 INVITE
  50. Server: Asterisk PBX 16.11.1
  51. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  52. Supported: replaces, timer
  53. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49c42999"
  54. Content-Length: 0
  55.  
  56.  
  57. <------------>
  58. Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  59.  
  60. <--- SIP read from UDP:192.168.10.229:5060 --->
  61. ACK sip:[email protected] SIP/2.0
  62. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ece9950c31089f;rport
  63. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  64. To: <sip:[email protected]>;tag=as440c3211
  65. Call-ID: [email protected]
  66. CSeq: 189 ACK
  67. Content-Length: 0
  68.  
  69. <------------->
  70. --- (7 headers 0 lines) ---
  71.  
  72. <--- SIP read from UDP:192.168.10.229:5060 --->
  73. INVITE sip:[email protected] SIP/2.0
  74. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ede9950c31089f;rport
  75. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  76. To: <sip:[email protected]>
  77. Call-ID: [email protected]
  78. CSeq: 190 INVITE
  79. Contact: <sip:[email protected]:5060>
  80. Authorization: Digest username="413", realm="asterisk", nonce="49c42999", uri="sip:[email protected]", response="a9073656c8efae7d035d355abfb53e29", algorithm=MD5
  81. Content-Type: application/sdp
  82. Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
  83. Max-Forwards: 70
  84. Supported: 100rel, replaces, from-change
  85. User-Agent: SIPPER for PhonerLite
  86. P-Preferred-Identity: <sip:[email protected]>
  87. Content-Length: 447
  88.  
  89. v=0
  90. o=- 1929311701 1 IN IP4 192.168.10.229
  91. s=SIPPER for PhonerLite
  92. c=IN IP4 192.168.10.229
  93. t=0 0
  94. m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 101
  95. a=rtpmap:107 opus/48000
  96. a=rtpmap:8 PCMA/8000
  97. a=rtpmap:0 PCMU/8000
  98. a=rtpmap:2 G726-32/8000
  99. a=rtpmap:3 GSM/8000
  100. a=rtpmap:97 iLBC/8000
  101. a=rtpmap:110 speex/8000
  102. a=rtpmap:111 speex/16000
  103. a=rtpmap:9 G722/8000
  104. a=rtpmap:101 telephone-event/8000
  105. a=fmtp:101 0-16
  106. a=ssrc:2188520094
  107. a=sendrecv
  108. <------------->
  109. --- (15 headers 19 lines) ---
  110. Sending to 192.168.10.229:5060 (no NAT)
  111. Using INVITE request as basis request - [email protected]
  112. Found peer '413' for '413' from 192.168.10.229:5060
  113. == Using SIP RTP CoS mark 5
  114. Got SDP version 1 and unique parts [- 1929311701 IN IP4 192.168.10.229]
  115. Found RTP audio format 107
  116. Found RTP audio format 8
  117. Found RTP audio format 0
  118. Found RTP audio format 2
  119. Found RTP audio format 3
  120. Found RTP audio format 97
  121. Found RTP audio format 110
  122. Found RTP audio format 111
  123. Found RTP audio format 9
  124. Found RTP audio format 101
  125. Found audio description format opus for ID 107
  126. Found audio description format PCMA for ID 8
  127. Found audio description format PCMU for ID 0
  128. Found audio description format G726-32 for ID 2
  129. Found audio description format GSM for ID 3
  130. Found audio description format iLBC for ID 97
  131. Found audio description format speex for ID 110
  132. Found audio description format speex for ID 111
  133. Found audio description format G722 for ID 9
  134. Found audio description format telephone-event for ID 101
  135. Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|g726|gsm|alaw|g722|ilbc|opus|speex|speex16)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
  136. Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
  137. Peer audio RTP is at port 192.168.10.229:5062
  138. Looking for 105 in from-internal (domain 192.168.10.227)
  139. sip_route_dump: route/path hop: <sip:[email protected]:5060>
  140.  
  141. <--- Transmitting (no NAT) to 192.168.10.229:5060 --->
  142. SIP/2.0 100 Trying
  143. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ede9950c31089f;received=192.168.10.229;rport=5060
  144. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  145. To: <sip:[email protected]>
  146. Call-ID: [email protected]
  147. CSeq: 190 INVITE
  148. Server: Asterisk PBX 16.11.1
  149. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  150. Supported: replaces, timer
  151. Contact: <sip:[email protected]:5060>
  152. Content-Length: 0
  153.  
  154.  
  155. <------------>
  156. -- Executing [105@from-internal:1] Dial("SIP/413-00000050", "OOH323/PanasMam/105,60,tTr") in new stack
  157. -- Called OOH323/PanasMam/105
  158.  
  159. <--- Transmitting (no NAT) to 192.168.10.229:5060 --->
  160. SIP/2.0 180 Ringing
  161. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ede9950c31089f;received=192.168.10.229;rport=5060
  162. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  163. To: <sip:[email protected]>;tag=as44798c50
  164. Call-ID: [email protected]
  165. CSeq: 190 INVITE
  166. Server: Asterisk PBX 16.11.1
  167. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  168. Supported: replaces, timer
  169. Contact: <sip:[email protected]:5060>
  170. Content-Length: 0
  171.  
  172.  
  173. <------------>
  174. == Everyone is busy/congested at this time (1:0/0/1)
  175. -- Auto fallthrough, channel 'SIP/413-00000050' status is 'CHANUNAVAIL'
  176.  
  177. <--- Reliably Transmitting (no NAT) to 192.168.10.229:5060 --->
  178. SIP/2.0 503 Service Unavailable
  179. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ede9950c31089f;received=192.168.10.229;rport=5060
  180. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  181. To: <sip:[email protected]>;tag=as44798c50
  182. Call-ID: [email protected]
  183. CSeq: 190 INVITE
  184. Server: Asterisk PBX 16.11.1
  185. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  186. Supported: replaces, timer
  187. X-Asterisk-HangupCause: Bearer capability not implemented
  188. X-Asterisk-HangupCauseCode: 65
  189. Content-Length: 0
  190.  
  191.  
  192. <------------>
  193.  
  194. <--- SIP read from UDP:192.168.10.229:5060 --->
  195. ACK sip:[email protected] SIP/2.0
  196. Via: SIP/2.0/UDP 192.168.10.229:5060;branch=z9hG4bK00f05160cec0ea1195ede9950c31089f;rport
  197. From: "PhonerLite" <sip:[email protected]>;tag=3494382261
  198. To: <sip:[email protected]>;tag=as44798c50
  199. Call-ID: [email protected]
  200. CSeq: 190 ACK
  201. Content-Length: 0
  202.  
  203. <------------->
  204. --- (7 headers 0 lines) ---
  205. Really destroying SIP dialog '[email protected]' Method: ACK
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