Advertisement

07.08.2008 at 05:21AM PDT, ID: 23546314
[x]
Attachment Details
[x]
The Solution Rating System

With so many solutions, how can you tell which solutions are most likely to help you and which ones are not? To provide you with a tool to use, we rate our solutions based on various elements that most accurately determine if a solution is a quality solution. To explain what factors affect the solution rating, here are the elements we take into consideration when formulating our solution rating.

  • The Grade of the Solution
  • The Zone Rank of the Expert Providing the Solution
  • The Number of Author and Expert Comments
  • The Number of Experts Contributing
  • The Feedback of the Community

Your Input Matters
Because of the way the system is set up, the most important variable in this equation is you. As a member of Experts Exchange, you are able to cast your vote on the quality of the solutions in regard to how complete, accurate, helpful and easy to understand each solution is. When you provide your feedback, each rating is adjusted accordingly. So, if you see a solution that has a poor rating that you think is a good solution, let us know by rating it. As you do, the rating will be adjusted and will become more accurate for other members of our site.

If you have any suggestions that you would like to make for our rating system, please ask a question in the Suggestions Zone of Community Support.

Thank you!

7.7

Problem with accepting a SIP trunk on IP 500

Asked by fzr600dave in IP PBX Systems, Voice Over IP, IP Telephony

Tags: , , ,

we have just setup a sip trunk with a friendly provider but despite the IPO receiving the SIP INVITE it doesn't accept the call, the logs suggest URI mapping but I cannot see whats missing.

There are many forums reporting similar issues none of which appear to be my scenario.
I enclose the sip debug info and my inbound routing screens

Licence is valid;
SipDebugInfo: License, Valid 1, Available 2, Consumed 0

my avaya is rejecting the call
    308529mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
    308529mS SipDebugInfo: Present Call, no match from URI in Request Line
    308529mS SipDebugInfo:  Sending code 404 to method INVITE

I have a static nat supplied throug a juniper netscreen firewall which is currently forwarding all ports to/from the ip office (on 192.168.40.50 to its public ip address)

Call is being directed to my sip user account (if registered) then my IP from an Asterisk PBX as a TRUNK.

asterisk config is;
[barwell_spain]
type=xxx
username=usnerma
secret=password
context=user
qualify=yes
defaultip=213.139.23.182
host=dynamic

Basically

a) Why is my ip office rejecting the call ?
b) why wont my IPO register on the sip server? (have tried register - i do not require STUN) but have set the IP to start STUN anyway, quite often the details change once I have set them.

At this stage i'm only concerned with inbound calls NOT outbound.

Running 4.1(12)
Start Free Trial
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:
308512mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
 
                     INVITE sip:442033551110@213.139.23.182 SIP/2.0
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                     To: <sip:442033551110@213.139.23.182>
                     Contact: <sip:08458620019@85.13.242.9>
                     Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                     CSeq: 102 INVITE
                     User-Agent: Asterisk PBX
                     Max-Forwards: 70
                     Date: Tue, 08 Jul 2008 13:12:41 GMT
                     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
                     Supported: replaces
                     Content-Type: application/sdp
                     Content-Length: 284
                     
                     v=0
                     o=root 53935 53935 IN IP4 85.13.242.9
                     s=session
                     c=IN IP4 85.13.242.9
                     t=0 0
                     m=audio 9612 RTP/AVP 8 3 0 101
                     a=rtpmap:8 PCMA/8000
                     a=rtpmap:3 GSM/8000
                     a=rtpmap:0 PCMU/8000
                     a=rtpmap:101 telephone-event/8000
                     a=fmtp:101 0-16
                     a=silenceSupp:off - - - -
                     a=ptime:20
                     a=sendrecv
    308513mS SIP Trunk: 10:Rx
                    INVITE sip:442033551110@213.139.23.182 SIP/2.0
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                    To: <sip:442033551110@213.139.23.182>
                    Contact: <sip:08458620019@85.13.242.9>
                    Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                    CSeq: 102 INVITE
                    User-Agent: Asterisk PBX
                    Max-Forwards: 70
                    Date: Tue, 08 Jul 2008 13:12:41 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
                    Supported: replaces
                    Content-Type: application/sdp
                    Content-Length: 284
                    
                    v=0
                    o=root 53935 53935 IN IP4 85.13.242.9
                    s=session
                    c=IN IP4 85.13.242.9
                    t=0 0
                    m=audio 9612 RTP/AVP 8 3 0 101
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:3 GSM/8000
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv
    308513mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget 
    308520mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1 
    308520mS SipDebugInfo: SIP: ProcessInbound Message
    308520mS SipDebugInfo:  Create Incoming EndPoint voip
    308521mS SipDebugInfo: License, Valid 1, Available 2, Consumed 0
    308521mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
    308522mS SipDebugInfo: ExtractCallIdFrom Message: Call Id value 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
    308522mS SipDebugInfo: ExtractCallerFromMessage: From Tag is as658ccd71
    308523mS SipDebugInfo: No Accept Header
    308524mS SipDebugInfo:  Sending code 100 to method INVITE 
    308524mS SipDebugInfo:  SendSIPResponse, Number of Tag Count, 0 
    308525mS SipDebugInfo: Sip_sendToNetwork packet of length 352
    308526mS SipDebugInfo: SIP Line (10): SendToTarget  550df209, 5060 
    308526mS SIP Trunk: 10:Tx
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                    To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
                    Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                    CSeq: 102 INVITE
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                    Content-Length: 0
                    
    308526mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
 
                     SIP/2.0 100 Trying
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                     To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
                     Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                     CSeq: 102 INVITE
                     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                     Content-Length: 0
                     
    308527mS SipDebugInfo: initialising mTxnContext
    308527mS SipDebugInfo: Present call entered 
    308528mS SipDebugInfo:  INVITE Received  ep fed54818, dialog fed14dcc 
    308528mS SipDebugInfo: ExtractRouteFromRecord, entered
    308528mS SipDebugInfo: ********************************************************* 
    308528mS SipDebugInfo: State Transtion form Old State 0 to New state 10 
    308528mS SipDebugInfo: ********************************************************* 
    308529mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
    308529mS SipDebugInfo: Present Call, no match from URI in Request Line 
    308529mS SipDebugInfo:  Sending code 404 to method INVITE 
    308530mS SipDebugInfo:  SendSIPResponse, Number of Tag Count, 1 
    308531mS SipDebugInfo: Sip_sendToNetwork packet of length 355
    308531mS SipDebugInfo: SIP Line (10): SendToTarget  550df209, 5060 
    308531mS SIP Trunk: 10:Tx
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                    To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
                    Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                    CSeq: 102 INVITE
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                    Content-Length: 0
                    
    308532mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
 
                     SIP/2.0 404 Not Found
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                     To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
                     Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                     CSeq: 102 INVITE
                     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                     Content-Length: 0
                     
    308533mS SipDebugInfo: ********************************************************* 
    308533mS SipDebugInfo: State Transtion form Old State 10 to New state 40 
    308533mS SipDebugInfo: ********************************************************* 
    308534mS SipDebugInfo: SIP Line (10): Freed Txn Key 2016
    308606mS RES: Tue 8/7/2008 13:59:59 FreeMem=47249756(34) CMMsg=4 (5) Buff=100 596 499 1052 3 Links=591
    308623mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
 
                     ACK sip:442033551110@213.139.23.182 SIP/2.0
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                     To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
                     Contact: <sip:08458620019@85.13.242.9>
                     Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                     CSeq: 102 ACK
                     User-Agent: Asterisk PBX
                     Max-Forwards: 70
                     Content-Length: 0
                     
    308624mS SIP Trunk: 10:Rx
                    ACK sip:442033551110@213.139.23.182 SIP/2.0
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK25354dd6;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as658ccd71
                    To: <sip:442033551110@213.139.23.182>;tag=106c369239e9cb5c
                    Contact: <sip:08458620019@85.13.242.9>
                    Call-ID: 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
                    CSeq: 102 ACK
                    User-Agent: Asterisk PBX
                    Max-Forwards: 70
                    Content-Length: 0
                    
    308624mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget 
    308628mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1 
    308628mS SipDebugInfo: SIP: ProcessInbound Message
    308628mS SipDebugInfo:  Find End Point 5b0dd2f97269b6e7538dac570a6dab15@netfuse.org
    308629mS SipDebugInfo: Process SIP request dialog fed14dcc, method ACK in Call state 40, Sdp state 0 
    308631mS SipDebugInfo: State is 40
    308632mS SipDebugInfo: SIP Line (10): Cannot free Txn Key 2015
    309631mS SipDebugInfo: EPTerminationTimeout, about to delete endpoint
    309631mS SipDebugInfo: SIPDialog destructor ... fed14dcc 
    309632mS SipDebugInfo:  Dialog destructor is deleting store d SIP message 
    309633mS SipDebugInfo: ~SipTrunkEndpoint
    309634mS CMTARGET:     0.1018.2 -1 BaseEP: ~CMTargetHandler
    313627mS SipDebugInfo: Timer 9 callback 
    314105mS RES: Tue 8/7/2008 14:00:04 FreeMem=47282144(20) CMMsg=4 (5) Buff=100 596 499 1052 3 Links=622
    315059mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
 
                     INVITE sip:442033551110@213.139.23.182 SIP/2.0
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                     To: <sip:442033551110@213.139.23.182>
                     Contact: <sip:08458620019@85.13.242.9>
                     Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                     CSeq: 102 INVITE
                     User-Agent: Asterisk PBX
                     Max-Forwards: 70
                     Date: Tue, 08 Jul 2008 13:12:48 GMT
                     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
                     Supported: replaces
                     Content-Type: application/sdp
                     Content-Length: 285
                     
                     v=0
                     o=root 53935 53935 IN IP4 85.13.242.9
                     s=session
                     c=IN IP4 85.13.242.9
                     t=0 0
                     m=audio 20364 RTP/AVP 8 3 0 101
                     a=rtpmap:8 PCMA/8000
                     a=rtpmap:3 GSM/8000
                     a=rtpmap:0 PCMU/8000
                     a=rtpmap:101 telephone-event/8000
                     a=fmtp:101 0-16
                     a=silenceSupp:off - - - -
                     a=ptime:20
                     a=sendrecv
    315059mS SIP Trunk: 10:Rx
                    INVITE sip:442033551110@213.139.23.182 SIP/2.0
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                    To: <sip:442033551110@213.139.23.182>
                    Contact: <sip:08458620019@85.13.242.9>
                    Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                    CSeq: 102 INVITE
                    User-Agent: Asterisk PBX
                    Max-Forwards: 70
                    Date: Tue, 08 Jul 2008 13:12:48 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
                    Supported: replaces
                    Content-Type: application/sdp
                    Content-Length: 285
                    
                    v=0
                    o=root 53935 53935 IN IP4 85.13.242.9
                    s=session
                    c=IN IP4 85.13.242.9
                    t=0 0
                    m=audio 20364 RTP/AVP 8 3 0 101
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:3 GSM/8000
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-16
                    a=silenceSupp:off - - - -
                    a=ptime:20
                    a=sendrecv
    315060mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget 
    315066mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1 
    315067mS SipDebugInfo: SIP: ProcessInbound Message
    315067mS SipDebugInfo:  Create Incoming EndPoint voip
    315068mS SipDebugInfo: License, Valid 1, Available 2, Consumed 0
    315068mS SipDebugInfo: CheckLineMonitors on SIP Endpoint - KEY & LAMP for SIP Trunk!
    315069mS SipDebugInfo: ExtractCallIdFrom Message: Call Id value 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
    315069mS SipDebugInfo: ExtractCallerFromMessage: From Tag is as52712853
    315069mS SipDebugInfo: No Accept Header
    315071mS SipDebugInfo:  Sending code 100 to method INVITE 
    315071mS SipDebugInfo:  SendSIPResponse, Number of Tag Count, 0 
    315072mS SipDebugInfo: Sip_sendToNetwork packet of length 352
    315072mS SipDebugInfo: SIP Line (10): SendToTarget  550df209, 5060 
    315072mS SIP Trunk: 10:Tx
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                    To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
                    Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                    CSeq: 102 INVITE
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                    Content-Length: 0
                    
    315073mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
 
                     SIP/2.0 100 Trying
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                     To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
                     Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                     CSeq: 102 INVITE
                     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                     Content-Length: 0
                     
    315073mS SipDebugInfo: initialising mTxnContext
    315074mS SipDebugInfo: Present call entered 
    315074mS SipDebugInfo:  INVITE Received  ep fed547ec, dialog fed14dcc 
    315074mS SipDebugInfo: ExtractRouteFromRecord, entered
    315075mS SipDebugInfo: ********************************************************* 
    315075mS SipDebugInfo: State Transtion form Old State 0 to New state 10 
    315075mS SipDebugInfo: ********************************************************* 
    315076mS SipDebugInfo: Present Call, no match (442033551110) from URI in To header.
    315076mS SipDebugInfo: Present Call, no match from URI in Request Line 
    315076mS SipDebugInfo:  Sending code 404 to method INVITE 
    315076mS SipDebugInfo:  SendSIPResponse, Number of Tag Count, 1 
    315078mS SipDebugInfo: Sip_sendToNetwork packet of length 355
    315078mS SipDebugInfo: SIP Line (10): SendToTarget  550df209, 5060 
    315078mS SIP Trunk: 10:Tx
                    SIP/2.0 404 Not Found
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                    To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
                    Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                    CSeq: 102 INVITE
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                    Content-Length: 0
                    
    315078mS SIP Tx: UDP 192.168.40.50:5060 -> 85.13.242.9:5060
 
                     SIP/2.0 404 Not Found
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                     To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
                     Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                     CSeq: 102 INVITE
                     Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
                     Content-Length: 0
                     
    315079mS SipDebugInfo: ********************************************************* 
    315080mS SipDebugInfo: State Transtion form Old State 10 to New state 40 
    315080mS SipDebugInfo: ********************************************************* 
    315080mS SipDebugInfo: SIP Line (10): Freed Txn Key 2016
    315169mS SIP Rx: UDP 85.13.242.9:5060 -> 192.168.40.50:5060
 
                     ACK sip:442033551110@213.139.23.182 SIP/2.0
                     Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                     From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                     To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
                     Contact: <sip:08458620019@85.13.242.9>
                     Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                     CSeq: 102 ACK
                     User-Agent: Asterisk PBX
                     Max-Forwards: 70
                     Content-Length: 0
                     
    315169mS SIP Trunk: 10:Rx
                    ACK sip:442033551110@213.139.23.182 SIP/2.0
                    Via: SIP/2.0/UDP 85.13.242.9:5060;branch=z9hG4bK1b472c77;rport
                    From: "08458620019" <sip:08458620019@netfuse.org>;tag=as52712853
                    To: <sip:442033551110@213.139.23.182>;tag=25372646c2d410f2
                    Contact: <sip:08458620019@85.13.242.9>
                    Call-ID: 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
                    CSeq: 102 ACK
                    User-Agent: Asterisk PBX
                    Max-Forwards: 70
                    Content-Length: 0
                    
    315170mS SipDebugInfo: MZ SIPDialog: ReceiveFromTarget 
    315173mS SipDebugInfo: MZ SIPDialog TXN : Decoding of message Succeded 1 
    315173mS SipDebugInfo: SIP: ProcessInbound Message
    315174mS SipDebugInfo:  Find End Point 7e463d9f3ca4a9e96a5618dd1e3c45ff@netfuse.org
    315175mS SipDebugInfo: Process SIP request dialog fed14dcc, method ACK in Call state 40, Sdp state 0 
    315177mS SipDebugInfo: State is 40
    315177mS SipDebugInfo: SIP Line (10): Cannot free Txn Key 2015
    316178mS SipDebugInfo: EPTerminationTimeout, about to delete endpoint
    316178mS SipDebugInfo: SIPDialog destructor ... fed14dcc 
    316178mS SipDebugInfo:  Dialog destructor is deleting store d SIP message 
    316179mS SipDebugInfo: ~SipTrunkEndpoint
Attachments:
 
avaya ip office system configuration - have tried openinternet etc
avaya ip office system configuration - have tried openinternet etc
 
 
sip line configuration - have tried registration required etc
sip line configuration - have tried registration required etc
 
 
sip URL - details are set to the inbound sip user
sip URL - details are set to the inbound sip user
 
[+][-]07.08.2008 at 12:42PM PDT, ID: 21957411

At Experts Exchange, members can ask their questions to thousands of technology professionals, also known as Experts. Experts compete and collaborate to answer those questions by leaving comments like this one.

Start your 7-day free trial to view this Expert Comment or ask the Experts your question.

 
[+][-]07.09.2008 at 01:58AM PDT, ID: 21961603

Often, when Experts are collaborating with members who have asked questions, they will request additional information about the problem. Askers respond with an author comment like this one.

Start your 7-day free trial to view this Author Comment or ask the Experts your question.

 
[+][-]07.09.2008 at 02:23AM PDT, ID: 21961710

Often, when Experts are collaborating with members who have asked questions, they will request additional information about the problem. Askers respond with an author comment like this one.

Start your 7-day free trial to view this Author Comment or ask the Experts your question.

 
[+][-]07.09.2008 at 02:59AM PDT, ID: 21961870

Often, when Experts are collaborating with members who have asked questions, they will request additional information about the problem. Askers respond with an author comment like this one.

Start your 7-day free trial to view this Author Comment or ask the Experts your question.

 
[+][-]07.09.2008 at 03:48PM PDT, ID: 21969060

View this solution now by starting your 7-day free trial. Setting up your free trial is quick, easy, and secure. We will return you to this solution, unlocked, when you're done.

 

About this solution

Zones: IP PBX Systems, Voice Over IP, IP Telephony
Tags: Avaya, IP Office, IP 500, inbound Sip Trunk
Sign Up Now!
Solution Provided By: fzr600dave
Participating Experts: 1
Solution Grade: B
 
 
[+][-]07.09.2008 at 04:02PM PDT, ID: 21969134

At Experts Exchange, members can ask their questions to thousands of technology professionals, also known as Experts. Experts compete and collaborate to answer those questions by leaving comments like this one.

Start your 7-day free trial to view this Expert Comment or ask the Experts your question.

 
[+][-]07.10.2008 at 03:32PM PDT, ID: 21977940

Experts Exchange has a courteous staff of administrators who help members get the most out of the website by means of administrative comments like this one.

Start your 7-day free trial to view this Administrative Comment or ask the Experts your question.

 
[+][-]07.13.2008 at 09:53PM PDT, ID: 21995736

Experts Exchange has a courteous staff of administrators who help members get the most out of the website by means of administrative comments like this one.

Start your 7-day free trial to view this Administrative Comment or ask the Experts your question.

 
[+][-]07.13.2008 at 09:53PM PDT, ID: 21995743

Experts Exchange has a courteous staff of administrators who help members get the most out of the website by means of administrative comments like this one.

Start your 7-day free trial to view this Administrative Comment or ask the Experts your question.

 
 
Loading Advertisement...
20080716-EE-VQP-32 / EE_QW_2_20070628