· 7 years ago · Nov 26, 2017, 01:56 AM
1[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
2<--- SIP read from UDP:192.168.1.158:5060 --->
3INVITE sip:5098556587@192.168.1.12;user=phone SIP/2.0
4Via: SIP/2.0/UDP 192.168.1.158;branch=z9hG4bK9f763194460023C5
5From: "204" <sip:204@192.168.1.12>;tag=F94F8C7A-CB1BBD3B
6To: <sip:5098556587@192.168.1.12;user=phone>
7CSeq: 1 INVITE
8Call-ID: 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
9Contact: <sip:204@192.168.1.158>
10Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
11User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.1.0933
12Accept-Language: en
13Supported: 100rel,replaces
14Allow-Events: talk,hold,conference
15Max-Forwards: 70
16Content-Type: application/sdp
17Content-Length: 296
18
19v=0
20o=- 1168133393 1168133393 IN IP4 192.168.1.158
21s=Polycom IP Phone
22c=IN IP4 192.168.1.158
23t=0 0
24a=sendrecv
25m=audio 2222 RTP/AVP 9 0 8 18 127
26a=rtpmap:9 G722/8000
27a=rtpmap:0 PCMU/8000
28a=rtpmap:8 PCMA/8000
29a=rtpmap:18 G729/8000
30a=fmtp:18 annexb=no
31a=rtpmap:127 telephone-event/8000
32<------------->
33[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: --- (15 headers 13 lines) ---
34[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Sending to 192.168.1.158:5060 (NAT)
35[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Using INVITE request as basis request - 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
36[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found peer '204' for '204' from 192.168.1.158:5060
37[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 9
38[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 0
39[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 8
40[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 18
41[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found RTP audio format 127
42[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format G722 for ID 9
43[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format PCMU for ID 0
44[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format PCMA for ID 8
45[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format G729 for ID 18
46[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Found audio description format telephone-event for ID 127
47[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
48[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
49[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Peer audio RTP is at port 192.168.1.158:2222
50[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Looking for 5098556587 in from-internal (domain 192.168.1.12)
51[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: list_route: hop: <sip:204@192.168.1.158>
52[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
53<--- Transmitting (no NAT) to 192.168.1.158:5060 --->
54SIP/2.0 100 Trying
55Via: SIP/2.0/UDP 192.168.1.158;branch=z9hG4bK9f763194460023C5;received=192.168.1.158
56From: "204" <sip:204@192.168.1.12>;tag=F94F8C7A-CB1BBD3B
57To: <sip:5098556587@192.168.1.12;user=phone>
58Call-ID: 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
59CSeq: 1 INVITE
60Server: FPBX-2.9.0(1.8.3.2)
61Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
62Supported: replaces, timer
63Contact: <sip:5098556587@192.168.1.12:5060>
64Content-Length: 0
65
66
67<------------>
68[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
69[2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Audio is at 5060
70[2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
71[2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding codec 0x8 (alaw) to SDP
72[2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding codec 0x2 (gsm) to SDP
73[2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
74[2011-09-14 10:21:16] VERBOSE[15184] chan_sip.c: Reliably Transmitting (NAT) to 64.2.142.29:5060:
75INVITE sip:5098556587@outbound.vitelity.net SIP/2.0
76Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK542211aa;rport
77Max-Forwards: 70
78From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
79To: <sip:5098556587@outbound.vitelity.net>
80Contact: <sip:njfragent@67.158.158.25:5060>
81Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
82CSeq: 102 INVITE
83User-Agent: FPBX-2.9.0(1.8.3.2)
84Date: Wed, 14 Sep 2011 17:21:16 GMT
85Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
86Supported: replaces, timer
87Remote-Party-ID: "Hvac corp" <sip:5097682249@67.158.158.25>;party=calling;privacy=off;screen=no
88Content-Type: application/sdp
89Content-Length: 287
90
91v=0
92o=root 1760196305 1760196305 IN IP4 67.158.158.25
93s=Asterisk PBX 1.8.3.2
94c=IN IP4 67.158.158.25
95t=0 0
96m=audio 11272 RTP/AVP 0 8 3 101
97a=rtpmap:0 PCMU/8000
98a=rtpmap:8 PCMA/8000
99a=rtpmap:3 GSM/8000
100a=rtpmap:101 telephone-event/8000
101a=fmtp:101 0-16
102a=ptime:20
103a=sendrecv
104
105---
106[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
107[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
108[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
109[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
110[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
111[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
112<--- SIP read from UDP:64.2.142.29:5060 --->
113SIP/2.0 407 Proxy Authentication Required
114Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK542211aa;received=67.158.158.25;rport=37041
115From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
116To: <sip:5098556587@outbound.vitelity.net>;tag=as02c943ec
117Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
118CSeq: 102 INVITE
119User-Agent: Asterisk PBX
120Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
121Supported: replaces
122Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="50fb97fb"
123Content-Length: 0
124
125<------------->
126[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: --- (11 headers 0 lines) ---
127[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Transmitting (NAT) to 64.2.142.29:5060:
128ACK sip:5098556587@outbound.vitelity.net SIP/2.0
129Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK542211aa;rport
130Max-Forwards: 70
131From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
132To: <sip:5098556587@outbound.vitelity.net>;tag=as02c943ec
133Contact: <sip:njfragent@67.158.158.25:5060>
134Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
135CSeq: 102 ACK
136User-Agent: FPBX-2.9.0(1.8.3.2)
137Content-Length: 0
138
139
140---
141[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Audio is at 5060
142[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
143[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding codec 0x8 (alaw) to SDP
144[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding codec 0x2 (gsm) to SDP
145[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
146[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c: Reliably Transmitting (NAT) to 64.2.142.29:5060:
147INVITE sip:5098556587@outbound.vitelity.net SIP/2.0
148Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK47af3aca;rport
149Max-Forwards: 70
150From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
151To: <sip:5098556587@outbound.vitelity.net>
152Contact: <sip:njfragent@67.158.158.25:5060>
153Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
154CSeq: 103 INVITE
155User-Agent: FPBX-2.9.0(1.8.3.2)
156Proxy-Authorization: Digest username="njfragent", realm="asterisk", algorithm=MD5, uri="sip:5098556587@outbound.vitelity.net", nonce="50fb97fb", response="dd061e97a3dfd2010b5c32d0f4d078b3"
157Date: Wed, 14 Sep 2011 17:21:16 GMT
158Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
159Supported: replaces, timer
160Remote-Party-ID: "Hvac corp" <sip:5097682249@67.158.158.25>;party=calling;privacy=off;screen=no
161Content-Type: application/sdp
162Content-Length: 287
163
164v=0
165o=root 1760196305 1760196306 IN IP4 67.158.158.25
166s=Asterisk PBX 1.8.3.2
167c=IN IP4 67.158.158.25
168t=0 0
169m=audio 11272 RTP/AVP 0 8 3 101
170a=rtpmap:0 PCMU/8000
171a=rtpmap:8 PCMA/8000
172a=rtpmap:3 GSM/8000
173a=rtpmap:101 telephone-event/8000
174a=fmtp:101 0-16
175a=ptime:20
176a=sendrecv
177
178---
179[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
180[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
181[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
182[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
183[2011-09-14 10:21:16] WARNING[2499] chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin)
184[2011-09-14 10:21:16] VERBOSE[3229] chan_sip.c:
185<--- SIP read from UDP:64.2.142.29:5060 --->
186SIP/2.0 100 Trying
187Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK47af3aca;received=67.158.158.25;rport=37041
188From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
189To: <sip:5098556587@outbound.vitelity.net>
190Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
191CSeq: 103 INVITE
192User-Agent: Asterisk PBX
193Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
194Supported: replaces
195Contact: <sip:5098556587@64.2.142.29>
196Content-Length: 0
197[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c:
198<--- SIP read from UDP:64.2.142.29:5060 --->
199SIP/2.0 183 Session Progress
200Via: SIP/2.0/UDP 67.158.158.25:5060;branch=z9hG4bK47af3aca;received=67.158.158.25;rport=37041
201From: "Hvac corp" <sip:njfragent@67.158.158.25>;tag=as76caf3ff
202To: <sip:5098556587@outbound.vitelity.net>;tag=as2fff8ffc
203Call-ID: 6ca7d90e51c09aff0aae49ae6d49a943@67.158.158.25:5060
204CSeq: 103 INVITE
205User-Agent: Asterisk PBX
206Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
207Supported: replaces
208Contact: <sip:5098556587@64.2.142.29>
209Content-Type: application/sdp
210Content-Length: 283
211
212v=0
213o=root 3697 3697 IN IP4 64.2.142.29
214s=session
215c=IN IP4 64.2.142.29
216t=0 0
217m=audio 12786 RTP/AVP 0 8 3 101
218a=rtpmap:0 PCMU/8000
219a=rtpmap:8 PCMA/8000
220a=rtpmap:3 GSM/8000
221a=rtpmap:101 telephone-event/8000
222a=fmtp:101 0-16
223a=silenceSupp:off - - - -
224a=ptime:20
225a=sendrecv
226<------------->
227[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: --- (12 headers 14 lines) ---
228[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 0
229[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 8
230[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 3
231[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found RTP audio format 101
232[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format PCMU for ID 0
233[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format PCMA for ID 8
234[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format GSM for ID 3
235[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Found audio description format telephone-event for ID 101
236[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
237[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
238[2011-09-14 10:21:18] VERBOSE[3229] chan_sip.c: Peer audio RTP is at port 64.2.142.29:12786
239[2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c: Audio is at 5060
240[2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
241[2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c: Adding codec 0x8 (alaw) to SDP
242[2011-09-14 10:21:18] VERBOSE[15184] chan_sip.c:
243<--- Transmitting (no NAT) to 192.168.1.158:5060 --->
244SIP/2.0 183 Session Progress
245Via: SIP/2.0/UDP 192.168.1.158;branch=z9hG4bK9f763194460023C5;received=192.168.1.158
246From: "204" <sip:204@192.168.1.12>;tag=F94F8C7A-CB1BBD3B
247To: <sip:5098556587@192.168.1.12;user=phone>;tag=as58e86aa0
248Call-ID: 9d13a40e-78b8a8af-b92b9de8@192.168.1.158
249CSeq: 1 INVITE
250Server: FPBX-2.9.0(1.8.3.2)
251Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
252Supported: replaces, timer
253Contact: <sip:5098556587@192.168.1.12:5060>
254Content-Type: application/sdp
255Content-Length: 204
256
257v=0
258o=root 1113813564 1113813564 IN IP4 192.168.1.12
259s=Asterisk PBX 1.8.3.2
260c=IN IP4 192.168.1.12
261t=0 0
262m=audio 19212 RTP/AVP 0 8
263a=rtpmap:0 PCMU/8000
264a=rtpmap:8 PCMA/8000
265a=ptime:20
266a=sendrecv