API tools faq
paste
Advertisement
Guest User

AASTRA 480i NO MUSIC ON HOLD ASTERISK 1.8.X

a guest
Mar 21st, 2011
99
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 10.83 KB | None | 0 0
raw download clone embed print report
  1. server100*CLI> sip set debug peer 103
  2. SIP Debugging Enabled for IP: 192.168.1.103
  3.  
  4. <--- SIP read from UDP:192.168.1.103:5060 --->
  5. INVITE sip:110@192.168.1.100:5060 SIP/2.0
  6. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKc67364c3e
  7. Max-Forwards: 70
  8. Content-Length: 226
  9. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  10. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  11. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  12. CSeq: 934584925 INVITE
  13. Supported: timer
  14. Allow-Events: talk,hold,conference
  15. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  16. Content-Type: application/sdp
  17. Contact: 9209821673 <sip:103@192.168.1.103:5060>
  18. Supported: replaces
  19. User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
  20.  
  21. v=0
  22. o=MxSIP 0 2112689786 IN IP4 192.168.1.103
  23. s=SIP Call
  24. c=IN IP4 0.0.0.0
  25. t=0 0
  26. m=audio 3000 RTP/AVP 0 101
  27. a=rtpmap:0 PCMU/8000
  28. a=rtpmap:101 telephone-event/8000
  29. a=fmtp:101 0-15
  30. a=ptime:20
  31. a=silenceSupp:off - - - -
  32.  
  33. <------------->
  34. --- (15 headers 11 lines) ---
  35. Sending to 192.168.1.103:5060 (no NAT)
  36. Found RTP audio format 0
  37. Found RTP audio format 101
  38. Found audio description format PCMU for ID 0
  39. Found audio description format telephone-event for ID 101
  40. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  41. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  42. Peer audio RTP is at port 0.0.0.0:3000
  43.  
  44. <--- Transmitting (no NAT) to 192.168.1.103:5060 --->
  45. SIP/2.0 100 Trying
  46. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKc67364c3e;received=192.168.1.103
  47. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  48. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  49. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  50. CSeq: 934584925 INVITE
  51. Server: Asterisk PBX 1.8.3
  52. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  53. Supported: replaces
  54. Contact: <sip:110@192.168.1.100:5060>
  55. Content-Length: 0
  56.  
  57.  
  58. <------------>
  59. Audio is at 5060
  60. Adding codec 0x4 (ulaw) to SDP
  61. Adding non-codec 0x1 (telephone-event) to SDP
  62.  
  63. <--- Reliably Transmitting (no NAT) to 192.168.1.103:5060 --->
  64. SIP/2.0 200 OK
  65. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKc67364c3e;received=192.168.1.103
  66. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  67. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  68. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  69. CSeq: 934584925 INVITE
  70. Server: Asterisk PBX 1.8.3
  71. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  72. Supported: replaces
  73. Contact: <sip:110@192.168.1.100:5060>
  74. Content-Type: application/sdp
  75. Content-Length: 261
  76.  
  77. v=0
  78. o=root 428767603 428767604 IN IP4 192.168.1.100
  79. s=Asterisk PBX 1.8.3
  80. c=IN IP4 192.168.1.100
  81. t=0 0
  82. m=audio 13522 RTP/AVP 0 101
  83. a=rtpmap:0 PCMU/8000
  84. a=rtpmap:101 telephone-event/8000
  85. a=fmtp:101 0-16
  86. a=silenceSupp:off - - - -
  87. a=ptime:20
  88. a=sendrecv
  89.  
  90. <------------>
  91.  
  92. <--- SIP read from UDP:192.168.1.103:5060 --->
  93. ACK sip:110@192.168.1.100:5060 SIP/2.0
  94. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKe65e3098f
  95. Max-Forwards: 70
  96. Content-Length: 0
  97. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  98. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  99. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  100. CSeq: 934584925 ACK
  101. Contact: 9209821673 <sip:103@192.168.1.103:5060>
  102. User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
  103.  
  104.  
  105. <------------->
  106. --- (10 headers 0 lines) ---
  107. == Spawn extension (internal, 0, 1) exited non-zero on 'SIP/vitel-inbound-000001dc'
  108. == Extension Changed 105[internal] new state Idle for Notify User 104
  109. == Extension Changed 105[internal] new state Idle for Notify User 101
  110. == Extension Changed 105[internal] new state Idle for Notify User 106
  111.  
  112. <--- SIP read from UDP:192.168.1.103:5060 --->
  113. REGISTER sip:192.168.1.100:5060 SIP/2.0
  114. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKcf1ff1eaa
  115. Max-Forwards: 70
  116. Content-Length: 0
  117. To: 9209821673 <sip:103@192.168.1.100:5060>
  118. From: 9209821673 <sip:103@192.168.1.100:5060>;tag=0caf0598dee49ca
  119. Call-ID: e82f614046c7f7a47047850079e6b8b8@192.168.1.103
  120. CSeq: 1542176036 REGISTER
  121. Contact: 9209821673 <sip:103@192.168.1.103:5060;transport=udp>
  122. Allow-Events: talk,hold,conference
  123. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  124. Authorization:Digest response="cd7b870a68df1a6a7300efd979bb6d67",username="103",realm="asterisk",nonce="784b846a",algorithm=MD5,uri="sip:192.168.1.100:5060"
  125. User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
  126.  
  127.  
  128. <------------->
  129. --- (13 headers 0 lines) ---
  130. Sending to 192.168.1.103:5060 (no NAT)
  131.  
  132. <--- Transmitting (no NAT) to 192.168.1.103:5060 --->
  133. SIP/2.0 401 Unauthorized
  134. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bKcf1ff1eaa;received=192.168.1.103
  135. From: 9209821673 <sip:103@192.168.1.100:5060>;tag=0caf0598dee49ca
  136. To: 9209821673 <sip:103@192.168.1.100:5060>;tag=as7850279e
  137. Call-ID: e82f614046c7f7a47047850079e6b8b8@192.168.1.103
  138. CSeq: 1542176036 REGISTER
  139. Server: Asterisk PBX 1.8.3
  140. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  141. Supported: replaces
  142. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="13592703"
  143. Content-Length: 0
  144.  
  145.  
  146. <------------>
  147. Scheduling destruction of SIP dialog 'e82f614046c7f7a47047850079e6b8b8@192.168.1.103' in 32000 ms (Method: REGISTER)
  148.  
  149. <--- SIP read from UDP:192.168.1.103:5060 --->
  150. REGISTER sip:192.168.1.100:5060 SIP/2.0
  151. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK7460fdd1c
  152. Max-Forwards: 70
  153. Content-Length: 0
  154. To: 9209821673 <sip:103@192.168.1.100:5060>
  155. From: 9209821673 <sip:103@192.168.1.100:5060>;tag=0caf0598dee49ca
  156. Call-ID: e82f614046c7f7a47047850079e6b8b8@192.168.1.103
  157. CSeq: 1542176037 REGISTER
  158. Contact: 9209821673 <sip:103@192.168.1.103:5060;transport=udp>
  159. Allow-Events: talk,hold,conference
  160. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  161. Authorization:Digest response="56d35894c32fde7d8a7d63acaf50d767",username="103",realm="asterisk",nonce="13592703",algorithm=MD5,uri="sip:192.168.1.100:5060"
  162. User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
  163.  
  164.  
  165. <------------->
  166. --- (13 headers 0 lines) ---
  167. Sending to 192.168.1.103:5060 (no NAT)
  168.  
  169. <--- Transmitting (no NAT) to 192.168.1.103:5060 --->
  170. SIP/2.0 200 OK
  171. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK7460fdd1c;received=192.168.1.103
  172. From: 9209821673 <sip:103@192.168.1.100:5060>;tag=0caf0598dee49ca
  173. To: 9209821673 <sip:103@192.168.1.100:5060>;tag=as7850279e
  174. Call-ID: e82f614046c7f7a47047850079e6b8b8@192.168.1.103
  175. CSeq: 1542176037 REGISTER
  176. Server: Asterisk PBX 1.8.3
  177. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  178. Supported: replaces
  179. Expires: 120
  180. Contact: <sip:103@192.168.1.103:5060;transport=udp>;expires=120
  181. Date: Mon, 21 Mar 2011 22:42:12 GMT
  182. Content-Length: 0
  183.  
  184.  
  185. <------------>
  186. Scheduling destruction of SIP dialog 'e82f614046c7f7a47047850079e6b8b8@192.168.1.103' in 32000 ms (Method: REGISTER)
  187.  
  188. <--- SIP read from UDP:192.168.1.103:5060 --->
  189. INVITE sip:110@192.168.1.100:5060 SIP/2.0
  190. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK43c4ae915
  191. Max-Forwards: 70
  192. Content-Length: 232
  193. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  194. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  195. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  196. CSeq: 934584926 INVITE
  197. Supported: timer
  198. Allow-Events: talk,hold,conference
  199. Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO
  200. Content-Type: application/sdp
  201. Contact: 9209821673 <sip:103@192.168.1.103:5060>
  202. Supported: replaces
  203. User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
  204.  
  205. v=0
  206. o=MxSIP 0 2112689787 IN IP4 192.168.1.103
  207. s=SIP Call
  208. c=IN IP4 192.168.1.103
  209. t=0 0
  210. m=audio 3000 RTP/AVP 0 101
  211. a=rtpmap:0 PCMU/8000
  212. a=rtpmap:101 telephone-event/8000
  213. a=fmtp:101 0-15
  214. a=ptime:20
  215. a=silenceSupp:off - - - -
  216.  
  217. <------------->
  218. --- (15 headers 11 lines) ---
  219. Sending to 192.168.1.103:5060 (no NAT)
  220. Found RTP audio format 0
  221. Found RTP audio format 101
  222. Found audio description format PCMU for ID 0
  223. Found audio description format telephone-event for ID 101
  224. Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  225. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  226. Peer audio RTP is at port 192.168.1.103:3000
  227.  
  228. <--- Transmitting (no NAT) to 192.168.1.103:5060 --->
  229. SIP/2.0 100 Trying
  230. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK43c4ae915;received=192.168.1.103
  231. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  232. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  233. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  234. CSeq: 934584926 INVITE
  235. Server: Asterisk PBX 1.8.3
  236. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  237. Supported: replaces
  238. Contact: <sip:110@192.168.1.100:5060>
  239. Content-Length: 0
  240.  
  241.  
  242. <------------>
  243. Audio is at 5060
  244. Adding codec 0x4 (ulaw) to SDP
  245. Adding non-codec 0x1 (telephone-event) to SDP
  246.  
  247. <--- Reliably Transmitting (no NAT) to 192.168.1.103:5060 --->
  248. SIP/2.0 200 OK
  249. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK43c4ae915;received=192.168.1.103
  250. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  251. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  252. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  253. CSeq: 934584926 INVITE
  254. Server: Asterisk PBX 1.8.3
  255. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  256. Supported: replaces
  257. Contact: <sip:110@192.168.1.100:5060>
  258. Content-Type: application/sdp
  259. Content-Length: 261
  260.  
  261. v=0
  262. o=root 428767603 428767605 IN IP4 192.168.1.100
  263. s=Asterisk PBX 1.8.3
  264. c=IN IP4 192.168.1.100
  265. t=0 0
  266. m=audio 13522 RTP/AVP 0 101
  267. a=rtpmap:0 PCMU/8000
  268. a=rtpmap:101 telephone-event/8000
  269. a=fmtp:101 0-16
  270. a=silenceSupp:off - - - -
  271. a=ptime:20
  272. a=sendrecv
  273.  
  274. <------------>
  275.  
  276. <--- SIP read from UDP:192.168.1.103:5060 --->
  277. ACK sip:110@192.168.1.100:5060 SIP/2.0
  278. Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK4cb922432
  279. Max-Forwards: 70
  280. Content-Length: 0
  281. To: "110" <sip:110@192.168.1.100>;tag=as164cf875
  282. From: <sip:103@192.168.1.103:5060>;tag=89d792e2b9325d4
  283. Call-ID: 3f8fec7c6ab45b1244e287034861a312@192.168.1.100:5060
  284. CSeq: 934584926 ACK
  285. Contact: 9209821673 <sip:103@192.168.1.103:5060>
  286. User-Agent: Aastra 480i/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45
  287.  
  288.  
  289. <------------->
  290. --- (10 headers 0 lines) ---
  291. server100*CLI> sip set debug off
  292. SIP Debugging Disabled
  293. == Extension Changed 103[internal] new state Idle for Notify User 105
  294. == Extension Changed 103[internal] new state Idle for Notify User 104
  295. == Extension Changed 103[internal] new state Idle for Notify User 101
  296. == Extension Changed 103[internal] new state Idle for Notify User 106
  297. == Spawn extension (outbound, 103, 5) exited non-zero on 'SIP/110-000001e8'
  298. server100*CLI>
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement
Public Pastes
Advertisement
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!