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發表於 2013-8-4 10:04 | 只看該作者

Asterisk 11.5.0 + NWT Internet NetTalk (依然不成功）

本帖最後由 角色 於 2013-12-27 01:33 編輯

因为有member要用Asterisk 11，但是怎样注册，只能打出，而不能打入。而我用Plain Asterisk 11.5.0，只能打入，而不能打出。（用Asterisk 1.8.22试过是没有问题的）

我的Asterisk sip.conf 和 extensions.conf configuration files

注意：

1、 由于我的NAS安装了好几个Asterisk servers，所以我把我的Asterisk 11放在 /opt/asterisk11的子目录下面。而port number用5080，我NAS的IP是10.0.88.6。

2、在/opt/asterisk11/etc/asterisk/rtp.conf，大家要根据自己的需要，可以作出修改，而在WAN router里的ports要作出应当的调整。

sip.conf contains
  1. [general]
  2. srvlookup = yes
  3. realm=hostname_of_your_wan_port
  4. externhost=hostname_of_your_wan_port
  5. fromdomain=hostname_of_your_wan_port
  6. address
  7. localnet=10.0.88.0/255.255.255.0 ;change it as per your Asterisk network address
  8. externrefresh = 1
  9. defaultexpirey=360
  10. bindport=5080
  11. bindaddr=10.0.88.6
  12. nat=force_rport,comedia
  13. qualify=yes
  14. disallow=all
  15. allow=ulaw,alaw,gsm
  16. alwaysauthreject=yes
  17. tos_sip=cs3
  18. tos_audio=ef
  19. tos_video=af41
  20. pedantic=yes
  21. context=front-desk


  24. register => 333445566:password@ngn2.nwtbb.com/333445566

  26. [nwt-nettalk]
  27. username=333445566
  28. type=peer
  29. secret=password
  30. qualify=yes
  31. port=5060
  32. nat=force_rport,comedia
  33. insecure = port,invite
  34. host=ngn2.nwtbb.com
  35. fromusername=333445566
  36. fromuser=333445566
  37. fromdomain=ngn2.nwtbb.com
  38. dtmfmode = rfc2833
  39. canreinvite = no
  40. defaultexpirey=300
  41. context = from-nwt-nettalk

  43. [2004]
  44. type=friend
  45. secret=password_for_2004
  46. qualify=yes
  47. host=dynamic
  48. nat=force_rport,comedia
  49. insecure=invite
  50. canreinvit=no
  51. context=internal
複製代碼
extensions.conf contains
  1. [front-desk]

  3. [internal]

  5. exten => _XXX.,1,Dial(SIP/${EXTEN}@nwt-nettalk,,r)
  6. exten => 2004,1,Dial(SIP/2004,,r)

  8. [from-nwt-nettalk]
  9. exten => 33445566,1,Dial(SIP/2004)
  10. exten => 33445566,2,Hangup
複製代碼
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角色

Rank: 2
2#
發表於 2013-8-4 10:10 | 只看該作者
本帖最後由 角色 於 2013-8-4 10:12 編輯

Outbound Call Simple Log
  1. TWTS-269PRO*CLI>
  2.   == Using SIP RTP TOS bits 184
  3.   == Using SIP RTP CoS mark 5
  4.     -- Executing [99663311@internal:1] Dial("SIP/2004-0000003c", "SIP/96331527@nwt-nettalk,,r") in new stack
  5.   == Using SIP RTP TOS bits 184
  6.   == Using SIP RTP CoS mark 5
  7.     -- Called SIP/99663311@nwt-nettalk
  8. [Aug  4 09:50:39] WARNING[15750][C-0000001f]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
  9. [Aug  4 09:50:39] WARNING[15750][C-0000001f]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
  10.     -- SIP/nwt-nettalk-0000003d is ringing
  11.     -- SIP/nwt-nettalk-0000003d is making progress passing it to SIP/2004-0000003c
  12. TWTS-269PRO*CLI>
  13. TWTS-269PRO*CLI>
複製代碼
..
..
我的手机响！我一接电话就出现下面的log
..
..
  1. TWTS-269PRO*CLI>
  2. TWTS-269PRO*CLI>
  3. [Aug  4 09:50:47] WARNING[15750][C-0000001f]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
  4. [Aug  4 09:50:47] WARNING[15750][C-0000001f]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
  5.     -- SIP/nwt-nettalk-0000003d answered SIP/2004-0000003c
  6.   == Spawn extension (internal, 99663311, 1) exited non-zero on 'SIP/2004-0000003c'
  7.     -- Got SIP response 500 "Server Internal Error" back from 203.176.254.198:5060
  8. TWTS-269PRO*CLI>
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角色

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3#
發表於 2013-8-4 10:16 | 只看該作者
Inbound （Incoming）call without any problem.
  1. TWTS-269PRO*CLI>
  2.   == Using SIP RTP TOS bits 184
  3.   == Using SIP RTP CoS mark 5
  4.     -- Executing [33445566@from-nwt-nettalk:1] Dial("SIP/nwt-nettalk-0000003e", "SIP/2004") in new stack
  5.   == Using SIP RTP TOS bits 184
  6.   == Using SIP RTP CoS mark 5
  7.     -- Called SIP/2004
  8.     -- SIP/2004-0000003f is ringing
  9. ..
  10. ..
  11. 我接电话后出现下面的信息：
  12. ..
  13. ..
  14.     -- SIP/2004-0000003f answered SIP/nwt-nettalk-0000003e
  15.     -- Locally bridging SIP/nwt-nettalk-0000003e and SIP/2004-0000003f
  16. ..
  17. ..
  18. 我说完收线
  19. ..
  20. ..
  21.   == Spawn extension (from-nwt-nettalk, 33445566, 1) exited non-zero on 'SIP/nwt-nettalk-0000003e'
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角色

Rank: 2
4#
發表於 2013-8-4 14:52 | 只看該作者
本帖最後由 角色 於 2013-8-4 14:59 編輯

Please note the following three posts describing the location of problem.
  1. TWTS-269PRO*CLI>
  2. TWTS-269PRO*CLI>
  3. TWTS-269PRO*CLI>
  4. TWTS-269PRO*CLI>
  5. TWTS-269PRO*CLI>
  6. TWTS-269PRO*CLI>

  8. <--- SIP read from UDP:10.0.88.22:5060 --->
  9. INVITE sip:99663311@10.0.88.6:5080 SIP/2.0
  10. Via: SIP/2.0/UDP 10.0.88.22:5060;rport;branch=z9hG4bKff918c1e86
  11. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  12. To: <sip:99663311@10.0.88.6:5080>
  13. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  14. Contact: <sip:2004@10.0.88.22:5060>
  15. CSeq: 1 INVITE
  16. Max-Forwards: 70
  17. Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE
  18. Supported: replaces
  19. Content-Type: application/sdp
  20. User-Agent: DGP306-O (1304100)
  21. Content-Length: 217

  23. v=0
  24. o=CMI-SIPUA 477 0 IN IP4 10.0.88.22
  25. s=SIP CALL
  26. c=IN IP4 10.0.88.22
  27. t=0 0
  28. m=audio 21864 RTP/AVP 0 8 4 18 101
  29. a=rtpmap:101 telephone-event/8000
  30. a=fmtp:101 0-15
  31. a=fmtp:18 annexb=no
  32. a=rtcp:21865
  33. a=sendrecv
  34. <------------->
  35. --- (13 headers 11 lines) ---
  36. Sending to 10.0.88.22:5060 (NAT)
  37. Sending to 10.0.88.22:5060 (NAT)
  38. Using INVITE request as basis request - [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  39. Found peer '2004' for '2004' from 10.0.88.22:5060
  40.   == Using SIP RTP TOS bits 184
  41.   == Using SIP RTP CoS mark 5
  42. Found RTP audio format 0
  43. Found RTP audio format 8
  44. Found RTP audio format 4
  45. Found RTP audio format 18
  46. Found RTP audio format 101
  47. Found audio description format telephone-event for ID 101
  48. Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
  49. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  50. Peer audio RTP is at port 10.0.88.22:21864
  51. Looking for 99663311 in internal (domain 10.0.88.6)
  52. list_route: hop: <sip:2004@10.0.88.22:5060>

  54. <--- Transmitting (NAT) to 10.0.88.22:5060 --->
  55. SIP/2.0 100 Trying
  56. Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
  57. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  58. To: <sip:99663311@10.0.88.6:5080>
  59. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  60. CSeq: 1 INVITE
  61. Server: Asterisk PBX 11.5.0
  62. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  63. Supported: replaces, timer
  64. Contact: <sip:99663311@10.0.88.6:5080>
  65. Content-Length: 0


  68. <------------>
  69.     -- Executing [99663311@internal:1] Dial("SIP/2004-00000042", "SIP/99663311@nwt-nettalk,,r") in new stack
  70.   == Using SIP RTP TOS bits 184
  71.   == Using SIP RTP CoS mark 5
  72. Audio is at 12134
  73. Adding codec 100003 (ulaw) to SDP
  74. Adding codec 100004 (alaw) to SDP
  75. Adding codec 100002 (gsm) to SDP
  76. Adding non-codec 0x1 (telephone-event) to SDP
  77. Reliably Transmitting (NAT) to 203.176.254.198:5060:
  78. INVITE sip:99663311@ngn2.nwtbb.com:5060 SIP/2.0
  79. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport
  80. Max-Forwards: 70
  81. From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  82. To: <sip:99663311@ngn2.nwtbb.com:5060>
  83. Contact: <sip:33445566@219.73.68.130:5080>
  84. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  85. CSeq: 102 INVITE
  86. User-Agent: Asterisk PBX 11.5.0
  87. Date: Sun, 04 Aug 2013 06:26:14 GMT
  88. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  89. Supported: replaces, timer
  90. Content-Type: application/sdp
  91. Content-Length: 311

  93. v=0
  94. o=root 1811917192 1811917192 IN IP4 219.73.68.130
  95. s=Asterisk PBX 11.5.0
  96. c=IN IP4 219.73.68.130
  97. t=0 0
  98. m=audio 12134 RTP/AVP 0 8 3 101
  99. a=rtpmap:0 PCMU/8000
  100. a=rtpmap:8 PCMA/8000
  101. a=rtpmap:3 GSM/8000
  102. a=rtpmap:101 telephone-event/8000
  103. a=fmtp:101 0-16
  104. a=silenceSupp:off - - - -
  105. a=ptime:20
  106. a=sendrecv

  108. ---
  109.     -- Called SIP/99663311@nwt-nettalk

  111. <--- Transmitting (NAT) to 10.0.88.22:5060 --->
  112. SIP/2.0 180 Ringing
  113. Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
  114. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  115. To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  116. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  117. CSeq: 1 INVITE
  118. Server: Asterisk PBX 11.5.0
  119. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  120. Supported: replaces, timer
  121. Contact: <sip:99663311@10.0.88.6:5080>
  122. Content-Length: 0


  125. <------------>

  127. <--- SIP read from UDP:203.176.254.198:5060 --->
  128. SIP/2.0 100 Trying
  129. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport=5080
  130. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  131. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  132. To: <sip:99663311@ngn2.nwtbb.com:5060>
  133. CSeq: 102 INVITE
  134. Content-Length: 0

  136. <------------->
  137. --- (7 headers 0 lines) ---

  139. <--- SIP read from UDP:203.176.254.198:5060 --->
  140. SIP/2.0 407 Proxy Authentication Required
  141. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport=5080
  142. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  143. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  144. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=hcxvyhoy
  145. CSeq: 102 INVITE
  146. Proxy-Authenticate: Digest realm="Huawei",nonce="14:25:51:59515",stale=false,algorithm=MD5
  147. Content-Length: 0

  149. <------------->
  150. --- (8 headers 0 lines) ---
  151. Transmitting (NAT) to 203.176.254.198:5060:
  152. ACK sip:99663311@ngn2.nwtbb.com:5060 SIP/2.0
  153. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK3fdc0df9;rport
  154. Max-Forwards: 70
  155. From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  156. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=hcxvyhoy
  157. Contact: <sip:33445566@219.73.68.130:5080>
  158. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  159. CSeq: 102 ACK
  160. User-Agent: Asterisk PBX 11.5.0
  161. Content-Length: 0


  164. ---
  165. Audio is at 12134
  166. Adding codec 100003 (ulaw) to SDP
  167. Adding codec 100004 (alaw) to SDP
  168. Adding codec 100002 (gsm) to SDP
  169. Adding non-codec 0x1 (telephone-event) to SDP
  170. Reliably Transmitting (NAT) to 203.176.254.198:5060:
  171. INVITE sip:99663311@ngn2.nwtbb.com:5060 SIP/2.0
  172. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport
  173. Max-Forwards: 70
  174. From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  175. To: <sip:99663311@ngn2.nwtbb.com:5060>
  176. Contact: <sip:33445566@219.73.68.130:5080>
  177. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  178. CSeq: 103 INVITE
  179. User-Agent: Asterisk PBX 11.5.0
  180. Proxy-Authorization: Digest username="33445566", realm="Huawei", algorithm=MD5, uri="sip:99663311@ngn2.nwtbb.com:5060", nonce="14:25:51:59515", response="e7c53f4651865f857b5f4d567752819e"
  181. Date: Sun, 04 Aug 2013 06:26:14 GMT
  182. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  183. Supported: replaces, timer
  184. Content-Type: application/sdp
  185. Content-Length: 311

  187. v=0
  188. o=root 1811917192 1811917193 IN IP4 219.73.68.130
  189. s=Asterisk PBX 11.5.0
  190. c=IN IP4 219.73.68.130
  191. t=0 0
  192. m=audio 12134 RTP/AVP 0 8 3 101
  193. a=rtpmap:0 PCMU/8000
  194. a=rtpmap:8 PCMA/8000
  195. a=rtpmap:3 GSM/8000
  196. a=rtpmap:101 telephone-event/8000
  197. a=fmtp:101 0-16
  198. a=silenceSupp:off - - - -
  199. a=ptime:20
  200. a=sendrecv
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角色

Rank: 2
5#
發表於 2013-8-4 14:53 | 只看該作者
本帖最後由 角色 於 2013-8-4 15:00 編輯
  1. <--- SIP read from UDP:203.176.254.198:5060 --->
  2. SIP/2.0 100 Trying
  3. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport=5080
  4. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  5. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  6. To: <sip:99663311@ngn2.nwtbb.com:5060>
  7. CSeq: 103 INVITE
  8. Content-Length: 0

  10. <------------->
  11. --- (7 headers 0 lines) ---

  13. <--- SIP read from UDP:203.176.254.198:5060 --->
  14. SIP/2.0 180 Ringing
  15. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport=5080
  16. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  17. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  18. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  19. CSeq: 103 INVITE
  20. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
  21. Contact: <sip:203.176.254.198:5060;user=phone>
  22. Content-Length: 301
  23. Content-Type: application/sdp

  25. v=0
  26. o=HuaweiSoftX3000 30838616 30838616 IN IP4 203.176.254.198
  27. s=Sip Call
  28. c=IN IP4 0.0.0.0
  29. t=0 0
  30. m=audio 0 RTP/AVP 0 8 101
  31. a=rtpmap:0 PCMU/8000
  32. a=rtpmap:8 PCMA/8000
  33. a=rtpmap:101 telephone-event/8000
  34. a=ptime:20
  35. a=silenceSupp:off - - - -
  36. a=ecan:fb on -
  37. a=X-fax
  38. a=fmtp:101 0-15
  39. a=inactive
  40. <------------->
  41. --- (10 headers 15 lines) ---
  42. list_route: hop: <sip:203.176.254.198:5060;user=phone>
  43. [Aug  4 14:26:18] WARNING[15750][C-00000022]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
  44. [Aug  4 14:26:18] WARNING[15750][C-00000022]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
  45.     -- SIP/nwt-nettalk-00000043 is ringing

  47. <--- Transmitting (NAT) to 10.0.88.22:5060 --->
  48. SIP/2.0 180 Ringing
  49. Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
  50. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  51. To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  52. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  53. CSeq: 1 INVITE
  54. Server: Asterisk PBX 11.5.0
  55. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  56. Supported: replaces, timer
  57. Contact: <sip:99663311@10.0.88.6:5080>
  58. Content-Length: 0


  61. <------------>
  62.     -- SIP/nwt-nettalk-00000043 is making progress passing it to SIP/2004-00000042
  63. Reliably Transmitting (NAT) to 203.176.254.198:5060:
  64. OPTIONS sip:ngn2.nwtbb.com SIP/2.0
  65. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK5095bc48;rport
  66. Max-Forwards: 70
  67. From: "asterisk" <sip:33445566@vpntw.homeftp.org>;tag=as69c8458c
  68. To: <sip:ngn2.nwtbb.com>
  69. Contact: <sip:33445566@219.73.68.130:5080>
  70. Call-ID: [email]46cfd6b96d486efd1ef09f185beb0fc2@vpntw.homeftp.org[/email]
  71. CSeq: 102 OPTIONS
  72. User-Agent: Asterisk PBX 11.5.0
  73. Date: Sun, 04 Aug 2013 06:26:21 GMT
  74. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  75. Supported: replaces, timer
  76. Content-Length: 0


  79. ---

  81. <--- SIP read from UDP:203.176.254.198:5060 --->
  82. SIP/2.0 200 OK
  83. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK5095bc48;rport=5080
  84. Call-ID: [email]46cfd6b96d486efd1ef09f185beb0fc2@vpntw.homeftp.org[/email]
  85. From: "asterisk"<sip:33445566@vpntw.homeftp.org>;tag=as69c8458c
  86. To: <sip:ngn2.nwtbb.com>;tag=culhbyoh
  87. CSeq: 102 OPTIONS
  88. Content-Length: 0

  90. <------------->
  91. --- (7 headers 0 lines) ---
  92. Really destroying SIP dialog '46cfd6b96d486efd1ef09f185beb0fc2@vpntw.homeftp.org' Method: OPTIONS
  93. TWTS-269PRO*CLI>
  94. TWTS-269PRO*CLI>
  95. TWTS-269PRO*CLI>
複製代碼
..
..
The called telephone rings
..
..
  1. TWTS-269PRO*CLI>
  2. TWTS-269PRO*CLI>
  3. TWTS-269PRO*CLI>
  4. TWTS-269PRO*CLI>
  5. TWTS-269PRO*CLI>
  6. TWTS-269PRO*CLI>
  7. TWTS-269PRO*CLI>
  8. TWTS-269PRO*CLI>
  9. TWTS-269PRO*CLI>
  10. TWTS-269PRO*CLI>
  11. TWTS-269PRO*CLI>
  12. TWTS-269PRO*CLI>
  13. TWTS-269PRO*CLI>
  14. TWTS-269PRO*CLI>
  15. TWTS-269PRO*CLI>
  16. TWTS-269PRO*CLI>
  17. TWTS-269PRO*CLI>
  18. Really destroying SIP dialog '5f3cf05319c609fe18ba0fea5f6ad854@10.0.88.22' Method: REGISTER

  20. <--- SIP read from UDP:203.176.254.198:5060 --->
  21. SIP/2.0 200 OK
  22. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK01c6bfea;rport=5080
  23. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  24. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  25. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  26. CSeq: 103 INVITE
  27. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
  28. Contact: <sip:203.176.254.198:5060;user=phone>
  29. Content-Length: 301
  30. Content-Type: application/sdp

  32. v=0
  33. o=HuaweiSoftX3000 30838616 30838617 IN IP4 203.176.254.198
  34. s=Sip Call
  35. c=IN IP4 0.0.0.0
  36. t=0 0
  37. m=audio 0 RTP/AVP 0 8 101
  38. a=rtpmap:0 PCMU/8000
  39. a=rtpmap:8 PCMA/8000
  40. a=rtpmap:101 telephone-event/8000
  41. a=ptime:20
  42. a=silenceSupp:off - - - -
  43. a=ecan:fb on -
  44. a=X-fax
  45. a=fmtp:101 0-15
  46. a=inactive
  47. <------------->
  48. --- (10 headers 15 lines) ---
  49. [Aug  4 14:26:33] WARNING[15750][C-00000022]: chan_sip.c:10092 process_sdp: Ignoring audio media offer because port number is zero
  50. [Aug  4 14:26:33] WARNING[15750][C-00000022]: chan_sip.c:10473 process_sdp: Failing due to no acceptable offer found
  51. list_route: hop: <sip:203.176.254.198:5060;user=phone>
  52. set_destination: Parsing <sip:203.176.254.198:5060;user=phone> for address/port to send to
  53. set_destination: set destination to 203.176.254.198:5060
  54. Transmitting (NAT) to 203.176.254.198:5060:
  55. ACK sip:203.176.254.198:5060;user=phone SIP/2.0
  56. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK734c65bc;rport
  57. Max-Forwards: 70
  58. From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  59. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  60. Contact: <sip:33445566@219.73.68.130:5080>
  61. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  62. CSeq: 103 ACK
  63. User-Agent: Asterisk PBX 11.5.0
  64. Content-Length: 0


  67. ---
  68. set_destination: Parsing <sip:203.176.254.198:5060;user=phone> for address/port to send to
  69. set_destination: set destination to 203.176.254.198:5060
  70. Reliably Transmitting (NAT) to 203.176.254.198:5060:
  71. BYE sip:203.176.254.198:5060;user=phone SIP/2.0
  72. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK392c49ed;rport
  73. Max-Forwards: 70
  74. From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  75. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  76. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  77. CSeq: 104 BYE
  78. User-Agent: Asterisk PBX 11.5.0
  79. Proxy-Authorization: Digest username="33445566", realm="Huawei", algorithm=MD5, uri="sip:203.176.254.198:5060", nonce="14:25:51:59515", response="19d6ae396e735c889d4a12b4db6b92b8"
  80. X-Asterisk-HangupCause: Unknown
  81. X-Asterisk-HangupCauseCode: 0
  82. Content-Length: 0
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角色

Rank: 2
6#
發表於 2013-8-4 14:53 | 只看該作者
本帖最後由 角色 於 2013-8-4 15:00 編輯
  1. ---
  2. Scheduling destruction of SIP dialog '2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com' in 6400 ms (Method: INVITE)
  3.     -- SIP/nwt-nettalk-00000043 answered SIP/2004-00000042
  4. Audio is at 12990
  5. Adding codec 100003 (ulaw) to SDP
  6. Adding codec 100004 (alaw) to SDP
  7. Adding non-codec 0x1 (telephone-event) to SDP

  9. <--- Reliably Transmitting (NAT) to 10.0.88.22:5060 --->
  10. SIP/2.0 200 OK
  11. Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
  12. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  13. To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  14. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  15. CSeq: 1 INVITE
  16. Server: Asterisk PBX 11.5.0
  17. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  18. Supported: replaces, timer
  19. Contact: <sip:99663311@10.0.88.6:5080>
  20. Content-Type: application/sdp
  21. Content-Length: 278

  23. v=0
  24. o=root 837645094 837645094 IN IP4 10.0.88.6
  25. s=Asterisk PBX 11.5.0
  26. c=IN IP4 10.0.88.6
  27. t=0 0
  28. m=audio 12990 RTP/AVP 0 8 101
  29. a=rtpmap:0 PCMU/8000
  30. a=rtpmap:8 PCMA/8000
  31. a=rtpmap:101 telephone-event/8000
  32. a=fmtp:101 0-16
  33. a=silenceSupp:off - - - -
  34. a=ptime:20
  35. a=sendrecv

  37. <------------>
  38. Scheduling destruction of SIP dialog '2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com' in 6400 ms (Method: INVITE)
  39. set_destination: Parsing <sip:203.176.254.198:5060;user=phone> for address/port to send to
  40. set_destination: set destination to 203.176.254.198:5060
  41. Reliably Transmitting (NAT) to 203.176.254.198:5060:
  42. BYE sip:203.176.254.198:5060;user=phone SIP/2.0
  43. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK20935ba0;rport
  44. Max-Forwards: 70
  45. From: "Eric_Ast-11.5" <sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  46. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  47. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  48. CSeq: 105 BYE
  49. User-Agent: Asterisk PBX 11.5.0
  50. Proxy-Authorization: Digest username="33445566", realm="Huawei", algorithm=MD5, uri="sip:203.176.254.198:5060", nonce="14:25:51:59515", response="19d6ae396e735c889d4a12b4db6b92b8"
  51. X-Asterisk-HangupCause: Normal Clearing
  52. X-Asterisk-HangupCauseCode: 16
  53. Content-Length: 0


  56. ---
  57.   == Spawn extension (internal, 99663311, 1) exited non-zero on 'SIP/2004-00000042'
  58. Scheduling destruction of SIP dialog '790da6076f0f53a7573d31e77297616f@10.0.88.22' in 6400 ms (Method: INVITE)

  60. <--- SIP read from UDP:203.176.254.198:5060 --->
  61. SIP/2.0 500 Server Internal Error
  62. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK20935ba0;rport=5080
  63. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  64. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  65. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  66. CSeq: 105 BYE
  67. Warning: 399 SE2000 "SSF00157L02501[5645] Unexpected message received"
  68. Content-Length: 0

  70. <------------->
  71. --- (8 headers 0 lines) ---
  72.     -- Got SIP response 500 "Server Internal Error" back from 203.176.254.198:5060

  74. <--- SIP read from UDP:203.176.254.198:5060 --->
  75. SIP/2.0 200 OK
  76. Via: SIP/2.0/UDP 219.73.68.130:5080;branch=z9hG4bK392c49ed;rport=5080
  77. Call-ID: [email]2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com[/email]
  78. From: "Eric_Ast-11.5"<sip:33445566@ngn2.nwtbb.com>;tag=as61ed4c9d
  79. To: <sip:99663311@ngn2.nwtbb.com:5060>;tag=yguxc3m3-CC-23
  80. CSeq: 104 BYE
  81. Content-Length: 0

  83. <------------->
  84. --- (7 headers 0 lines) ---
  85. Really destroying SIP dialog '2504a5517f8b28f432e2a79a554ebc7e@ngn2.nwtbb.com' Method: INVITE
  86. Retransmitting #1 (NAT) to 10.0.88.22:5060:
  87. SIP/2.0 200 OK
  88. Via: SIP/2.0/UDP 10.0.88.22:5060;branch=z9hG4bKff918c1e86;received=10.0.88.22;rport=5060
  89. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  90. To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  91. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  92. CSeq: 1 INVITE
  93. Server: Asterisk PBX 11.5.0
  94. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  95. Supported: replaces, timer
  96. Contact: <sip:99663311@10.0.88.6:5080>
  97. Content-Type: application/sdp
  98. Content-Length: 278

  100. v=0
  101. o=root 837645094 837645094 IN IP4 10.0.88.6
  102. s=Asterisk PBX 11.5.0
  103. c=IN IP4 10.0.88.6
  104. t=0 0
  105. m=audio 12990 RTP/AVP 0 8 101
  106. a=rtpmap:0 PCMU/8000
  107. a=rtpmap:8 PCMA/8000
  108. a=rtpmap:101 telephone-event/8000
  109. a=fmtp:101 0-16
  110. a=silenceSupp:off - - - -
  111. a=ptime:20
  112. a=sendrecv

  114. ---

  116. <--- SIP read from UDP:10.0.88.22:5060 --->
  117. ACK sip:99663311@10.0.88.6:5080 SIP/2.0
  118. Via: SIP/2.0/UDP 10.0.88.22:5060;rport;branch=z9hG4bKba10cf7ffe
  119. From: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  120. To: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  121. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  122. Contact: <sip:2004@10.0.88.22:5060>
  123. CSeq: 1 ACK
  124. Max-Forwards: 70
  125. Content-Length: 0

  127. <------------->
  128. --- (9 headers 0 lines) ---
  129. set_destination: Parsing <sip:2004@10.0.88.22:5060> for address/port to send to
  130. set_destination: set destination to 10.0.88.22:5060
  131. Reliably Transmitting (NAT) to 10.0.88.22:5060:
  132. BYE sip:2004@10.0.88.22:5060 SIP/2.0
  133. Via: SIP/2.0/UDP 10.0.88.6:5080;branch=z9hG4bK1bbfa85a;rport
  134. Max-Forwards: 70
  135. From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  136. To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  137. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  138. CSeq: 102 BYE
  139. User-Agent: Asterisk PBX 11.5.0
  140. X-Asterisk-HangupCause: Normal Clearing
  141. X-Asterisk-HangupCauseCode: 16
  142. Content-Length: 0


  145. ---
  146. Scheduling destruction of SIP dialog '790da6076f0f53a7573d31e77297616f@10.0.88.22' in 6400 ms (Method: ACK)
  147. Retransmitting #1 (NAT) to 10.0.88.22:5060:
  148. BYE sip:2004@10.0.88.22:5060 SIP/2.0
  149. Via: SIP/2.0/UDP 10.0.88.6:5080;branch=z9hG4bK1bbfa85a;rport
  150. Max-Forwards: 70
  151. From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  152. To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  153. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  154. CSeq: 102 BYE
  155. User-Agent: Asterisk PBX 11.5.0
  156. X-Asterisk-HangupCause: Normal Clearing
  157. X-Asterisk-HangupCauseCode: 16
  158. Content-Length: 0


  161. ---

  163. <--- SIP read from UDP:10.0.88.22:5060 --->
  164. SIP/2.0 200 OK
  165. Via: SIP/2.0/UDP 10.0.88.6:5080;rport=5080;received=10.0.88.6;branch=z9hG4bK1bbfa85a
  166. From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  167. To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  168. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  169. CSeq: 102 BYE
  170. Content-Length: 0

  172. <------------->
  173. --- (7 headers 0 lines) ---
  174. SIP Response message for INCOMING dialog BYE arrived
  175. Really destroying SIP dialog '790da6076f0f53a7573d31e77297616f@10.0.88.22' Method: ACK

  177. <--- SIP read from UDP:10.0.88.22:5060 --->
  178. SIP/2.0 200 OK
  179. Via: SIP/2.0/UDP 10.0.88.6:5080;rport=5080;received=10.0.88.6;branch=z9hG4bK1bbfa85a
  180. From: <sip:99663311@10.0.88.6:5080>;tag=as512c33a1
  181. To: "Eric_Ast-11.5" <sip:2004@10.0.88.6:5080>;tag=307df92f
  182. Call-ID: [email]790da6076f0f53a7573d31e77297616f@10.0.88.22[/email]
  183. CSeq: 102 BYE
  184. Content-Length: 0

  186. <------------->
  187. --- (7 headers 0 lines) ---
  188. TWTS-269PRO*CLI>
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