Copy into clipboard    Syntax: S, added 110 days ago


52C971cE   Copy into clipboard

14074 characters | 372 lines | 104.0 kb
raw | download 
  1
  2
  3
  4
  5
  6
  7
  8
  9
 10
 11
 12
 13
 14
 15
 16
 17
 18
 19
 20
 21
 22
 23
 24
 25
 26
 27
 28
 29
 30
 31
 32
 33
 34
 35
 36
 37
 38
 39
 40
 41
 42
 43
 44
 45
 46
 47
 48
 49
 50
 51
 52
 53
 54
 55
 56
 57
 58
 59
 60
 61
 62
 63
 64
 65
 66
 67
 68
 69
 70
 71
 72
 73
 74
 75
 76
 77
 78
 79
 80
 81
 82
 83
 84
 85
 86
 87
 88
 89
 90
 91
 92
 93
 94
 95
 96
 97
 98
 99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
router*CLI> 
router*CLI> 
router*CLI> 
Really destroying SIP dialog '5f7375b479e27b504c1ae14e03bfe5b0@marcusbrutus.soho.on.net' Method: OPTIONS
Really destroying SIP dialog '24accf7e09537a740edcc4ce7d51387e@marcusbrutus.soho.on.net' Method: OPTIONS
Really destroying SIP dialog '64acce4b6b22708c5f4943416b5e7a70@marcusbrutus.soho.on.net' Method: REGISTER

<--- SIP read from UDP:192.168.0.64:5080 --->
INVITE sip:4321@router.lan SIP/2.0
Via: SIP/2.0/UDP 192.168.0.64:5080;rport;branch=z9hG4bKZSStSD3Nt90ZF
Max-Forwards: 69
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611175 INVITE
Contact: <sip:gw+router@192.168.0.64:5080;transport=udp;gw=router>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f697e5a 2011-08-29 13-28-02 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 205
X-FS-Support: update_display
Remote-Party-ID: "196907" <sip:196907@router.lan>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1327864005 1327864006 IN IP4 192.168.0.64
s=FreeSWITCH
c=IN IP4 192.168.0.64
t=0 0
m=audio 30474 RTP/AVP 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (17 headers 9 lines) ---
Sending to 192.168.0.64:5080 (NAT)
Using INVITE request as basis request - 2e1d2572-c596-122f-fa9e-f46d04909629
Found peer '19691' for '19691' from 192.168.0.64:5080

<--- Reliably Transmitting (NAT) to 192.168.0.64:5080 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.64:5080;branch=z9hG4bKZSStSD3Nt90ZF;received=192.168.0.64;rport=5080
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>;tag=as560c3a29
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611175 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a99d417"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2e1d2572-c596-122f-fa9e-f46d04909629' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.64:5080 --->
ACK sip:4321@router.lan SIP/2.0
Via: SIP/2.0/UDP 192.168.0.64:5080;rport;branch=z9hG4bKZSStSD3Nt90ZF
Max-Forwards: 69
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>;tag=as560c3a29
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611175 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.64:5080 --->
INVITE sip:4321@router.lan SIP/2.0
Via: SIP/2.0/UDP 192.168.0.64:5080;rport;branch=z9hG4bK02jKU8KSQjQjB
Max-Forwards: 69
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611176 INVITE
Contact: <sip:gw+router@192.168.0.64:5080;transport=udp;gw=router>
Expires: 600
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f697e5a 2011-08-29 13-28-02 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Authorization: Digest username="19691", realm="asterisk", nonce="6a99d417", algorithm=MD5, uri="sip:4321@router.lan", response="f5d1ea930b6f3f3bb5fe14c93e854cb0"
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 205
X-FS-Support: update_display
Remote-Party-ID: "196907" <sip:196907@router.lan>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1327864005 1327864006 IN IP4 192.168.0.64
s=FreeSWITCH
c=IN IP4 192.168.0.64
t=0 0
m=audio 30474 RTP/AVP 0 8 3 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (19 headers 9 lines) ---
Sending to 192.168.0.64:5080 (NAT)
Using INVITE request as basis request - 2e1d2572-c596-122f-fa9e-f46d04909629
Found peer '19691' for '19691' from 192.168.0.64:5080
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 13
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN|), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.64:30474
Looking for 4321 in context-user-19691 (domain router.lan)
list_route: hop: <sip:gw+router@192.168.0.64:5080;transport=udp;gw=router>

<--- Transmitting (NAT) to 192.168.0.64:5080 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.64:5080;branch=z9hG4bK02jKU8KSQjQjB;received=192.168.0.64;rport=5080
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611176 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:4321@192.168.0.1:5060>
Content-Length: 0


<------------>
    -- Executing [4321@context-user-19691:1] Dial("SIP/19691-00000006", "SIP/4321@peer-196907_conf_freedoh_net,60,r") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 173.77.234.116:5080:
INVITE sip:4321@conf.freedoh.net:5080 SIP/2.0
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK07e3a271;rport
Max-Forwards: 70
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>
Contact: <sip:196907@150.101.108.88:5060>
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 INVITE
User-Agent: PBX
Date: Mon, 30 Jan 2012 03:34:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1126628916 1126628916 IN IP4 150.101.108.88
s=Asterisk PBX 1.8.8.0
c=IN IP4 150.101.108.88
t=0 0
m=audio 19882 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called SIP/4321@peer-196907_conf_freedoh_net

<--- Transmitting (NAT) to 192.168.0.64:5080 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.64:5080;branch=z9hG4bK02jKU8KSQjQjB;received=192.168.0.64;rport=5080
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>;tag=as4bc052a0
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611176 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:4321@192.168.0.1:5060>
Content-Length: 0


<------------>
Retransmitting #1 (NAT) to 173.77.234.116:5080:
INVITE sip:4321@conf.freedoh.net:5080 SIP/2.0
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK07e3a271;rport
Max-Forwards: 70
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>
Contact: <sip:196907@150.101.108.88:5060>
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 INVITE
User-Agent: PBX
Date: Mon, 30 Jan 2012 03:34:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1126628916 1126628916 IN IP4 150.101.108.88
s=Asterisk PBX 1.8.8.0
c=IN IP4 150.101.108.88
t=0 0
m=audio 19882 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:173.77.234.116:5080 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK07e3a271;rport=5060
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0a503b1 2012-01-18 18-08-52 -0600
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:173.77.234.116:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK07e3a271;rport=5060
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>;tag=j5c7X5Q4X33rH
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 INVITE
Contact: <sip:4321@192.168.1.131:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0a503b1 2012-01-18 18-08-52 -0600
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 190
Remote-Party-ID: "Outbound Call" <sip:4321@conf.freedoh.net>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1043594931 1043594932 IN IP4 192.168.1.131
s=FreeSWITCH
c=IN IP4 192.168.1.131
t=0 0
m=audio 19232 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (16 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.131:19232
list_route: hop: <sip:4321@192.168.1.131:5080;transport=udp>
set_destination: Parsing <sip:4321@192.168.1.131:5080;transport=udp> for address/port to send to
set_destination: set destination to 192.168.1.131:5080
Transmitting (NAT) to 173.77.234.116:5080:
ACK sip:4321@192.168.1.131:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK6265b41d;rport
Max-Forwards: 70
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>;tag=j5c7X5Q4X33rH
Contact: <sip:196907@150.101.108.88:5060>
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0


---
    -- SIP/peer-196907_conf_freedoh_net-00000007 answered SIP/19691-00000006
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 192.168.0.64:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.64:5080;branch=z9hG4bK02jKU8KSQjQjB;received=192.168.0.64;rport=5080
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>;tag=as4bc052a0
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611176 INVITE
Server: PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:4321@192.168.0.1:5060>
Content-Type: application/sdp
Content-Length: 229

v=0
o=root 1270204476 1270204476 IN IP4 192.168.0.1
s=Asterisk PBX 1.8.8.0
c=IN IP4 192.168.0.1
t=0 0
m=audio 19852 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/19691-00000006 and SIP/peer-196907_conf_freedoh_net-00000007

<--- SIP read from UDP:192.168.0.64:5080 --->
ACK sip:4321@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.64:5080;rport;branch=z9hG4bK1BccX34vmUD5p
Max-Forwards: 70
From: "196907" <sip:19691@router.lan;transport=udp>;tag=S4UUSpm0U8v6r
To: <sip:4321@router.lan>;tag=as4bc052a0
Call-ID: 2e1d2572-c596-122f-fa9e-f46d04909629
CSeq: 23611176 ACK
Contact: <sip:gw+router@192.168.0.64:5080;transport=udp;gw=router>
Authorization: Digest username="19691", realm="asterisk", nonce="6a99d417", algorithm=MD5, uri="sip:4321@router.lan", response="f5d1ea930b6f3f3bb5fe14c93e854cb0"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:173.77.234.116:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK07e3a271;rport=5060
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>;tag=j5c7X5Q4X33rH
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 INVITE
Contact: <sip:4321@192.168.1.131:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-0a503b1 2012-01-18 18-08-52 -0600
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 190
Remote-Party-ID: "Outbound Call" <sip:4321@conf.freedoh.net>;party=calling;privacy=off;screen=no

v=0
o=FreeSWITCH 1043594931 1043594932 IN IP4 192.168.1.131
s=FreeSWITCH
c=IN IP4 192.168.1.131
t=0 0
m=audio 19232 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
<------------->
--- (16 headers 9 lines) ---
set_destination: Parsing <sip:4321@192.168.1.131:5080;transport=udp> for address/port to send to
set_destination: set destination to 192.168.1.131:5080
Transmitting (NAT) to 173.77.234.116:5080:
ACK sip:4321@192.168.1.131:5080;transport=udp SIP/2.0
Via: SIP/2.0/UDP 150.101.108.88:5060;branch=z9hG4bK285b20fd;rport
Max-Forwards: 70
From: "pbx64" <sip:196907@conf.freedoh.net>;tag=as188c84a2
To: <sip:4321@conf.freedoh.net:5080>;tag=j5c7X5Q4X33rH
Contact: <sip:196907@150.101.108.88:5060>
Call-ID: 4ffe2eea4246a0f0751dc4c1114e9846@conf.freedoh.net
CSeq: 102 ACK
User-Agent: PBX
Content-Length: 0