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mfitzgerald
Vetran
Joined: June 14 2006
Location: United States
Posts: 142
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| Posted: September 05 2008 at 6:06pm | IP Logged
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We have experienced some odd behavior with LanScapes handling of a 480 message.
Our application has code to auto-answer an incoming call. At the time we were still learning how to incorporate the LS Media Engine (5.12.3.30) into our app, so the implementation may not have been the best. After reviewing the method in which the SingleLinePhone App does this (recently LS 5.12.8.9), I altered the application to simulate this.
Ex:
Early Implementation:
* On SipIncomingCallInitialized event
* If the app was available we start a thread that would wait for the phone lines status SipOkToAnswerCall before Going Off Hook.
* If the app was not available we would attempt to send a 480.
note: because the phone line had not received a SipOkToAnswerCall we could not use the AbortIncomingCall() function.
* Instead we used the IncomingCallInitEventData and set TerminateIncomingCall = TRUE and SipResponseCode to 480.
Update (test):
The code was modified to use the AbortIncomingCall() function after the phone line status was SipOkToAnswerCall
Regardless of the implementation the results were largely the same.
The Issue:
When we receive a call we do not want to take at this time, we do not want to take it at that time. We do however want to take that call as soon as the application is available.
NOTE: The switch will continue to send the INVITE requests for a phone call even though it received the 480 message. We noticed LS did not respond to any further INVITES from that call, however once the app was made available we really did not seem to have a way to tell LS to let the call come through. The INVITES (for that call) were still ignored.
NOTE: Any INVITES from another call were allowed.
This is not a desired effect. We want the person who has been waiting the longest to get their call to come through.
When we tested with a different softphone we noticed that even though the Switch would continue to send INVITE requests for the 480-ed call, it would ALWAYS respond with the 480 message for all INVITES until their phone was made available.
Note: Another ill effect we noticed, was using the method with the IncomingCallInitEventData. LS would send a 481 when the Switch would send a CANCEL for subsequent INVITES that LS did not respond to. It appears as though LS is not keeping track of the CSeq numbers and does not put these messages in the SIP stack (I am assuming here). However, when using the AbortIncomingCall() function there were no 481 messages. However, LS never responded to the INVITES and the switch still sent CANCEL messages.
I have tested and think the 486 solution via BusyOutLine() would resolve these issues. However we do not have official confirmation of the Switch providers support for 486. (No offence to them but they did seem to take a while to officially support 480). I tend to prefer 486 method, but until the provider officially supports it, it is not a good idea to assume support will continue.
At this time I am working on getting the 486 solution to work, it is somewhat more involved, but it does appear to have the desired effect of answering the 486-ed call as soon as the app is available.
Questions
Something else I noted. LS did at times allow the 480-ed call to come through and allow the app to answer the call after roughly 3 minutes or so--Provided the switch did not given up and terminate the call. This was not always the case, but it did occur on occasion. It appears there is some time limit that LS will Ignore a call. If this is the case, is there some way to tell LS to accept calls, or modify or remove the time limit?
Why did LS not respond to subsequent INVITES for 480-ed calls?
Why did LS not seem to understand the CANCEL request from the Switch for a timed-out INVITE (i.e. LS sent 481).
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 09 2008 at 9:07am | IP Logged
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Hi Fitz,
To start, good explanation of your situation. The good news is, we are sure we can perform an update to the media engine to get your call flows behaving exactly as you would expect.
You >>>
NOTE: The switch will continue to send the INVITE requests for a phone call even though it received the 480 message. We noticed LS did not respond to any further INVITES from that call, however once the app was made available we really did not seem to have a way to tell LS to let the call come through. The INVITES (for that call) were still ignored.
<<< Support
We understand what you are describing. We need to see the SIP for the secondary calls (INVITES) the switch is initiating. This sounds like a very simple change in the media engine. It sounds like we need to augment the default behavior of the media engine when it is instructed to “ignore” an incoming call. Does not sound like a big deal.
We would at least expect the switch to change the ‘To:” SIP header tags in INVITE requests that are being retried – note that the call Ids for retried calls may still be the same. In this case, the media engine should allow secondary calls using the same call ID but having changed “To:” header tags. Its considered a new call dialog. If we can get the SIP log for this scenario from you, then we will look into it today and see what’ up. It should be simple to resolve.
You >>>
When we tested with a different softphone we noticed that even though the Switch would continue to send INVITE requests for the 480-ed call, it would ALWAYS respond with the 480 message for all INVITES until their phone was made available.
<<< Support
Understood. The media engine should also behave in this way. If we look at the SIP for the INVITES coming from the switch, we should be able to see why the media engine is not behaving the same way.
You >>>
Something else I noted. LS did at times allow the 480-ed call to come through and allow the app to answer the call after roughly 3 minutes or so--Provided the switch did not given up and terminate the call. This was not always the case, but it did occur on occasion. It appears there is some time limit that LS will Ignore a call. If this is the case, is there some way to tell LS to accept calls, or modify or remove the time limit?
<<< Support
Internally, the media engine maintains a cache of call dialogs for a certain amount of time. Its just an implementation issue on how the media engine behaves when it is told to ignore a call. You are exactly correct when you state that there is a “time limit” on a call dialog. We may be able to remove the time limit all together when an app tells the media engine to “ignore” a call – but we will need to see the SIP for the INVITES to determine how to best handle this situation properly and without hacks.
You >>>
Why did LS not respond to subsequent INVITES for 480-ed calls?
<<< Support
Not sure. For new call dialogs, the media engine should at least be looking at the call ID and the “To:” header’s tag value. If the switch is using the same call Ids and the “To:” header tag value for retried INVITE requests that are being received by the media engine, then that may be the issue. The SIP log will tell us what is occurring. Regardless of the situation, we can implement the proper change in the media engine to get us the correct behavior. This is why having a support agreement between us is so very important. We can “cut” new product images very quickly in most cases to over come SIP inter-op issues.
You >>>
Why did LS not seem to understand the CANCEL request from the Switch for a timed-out INVITE (i.e. LS sent 481).
<<< Support
Good question. This would normally occur if the call dialog for the CANCEL could not be matched up with an existing call dialog the media engine knows about. The call ID and “To:” header tag in the CANCEL should match the call ID and “To:” header tag in a previously received INVITE request. If the CANCEL comes in after the call dialog cache in the media engine has been “cleaned up”, then I would expect the “481 Transaction Does Not Exist” to be sent by the media engine. We can look at the Sip for this too to see what is going on.
Fitz, please post the SIP log for retried INVITES coming from the switch (that were previously “ignored” by the media engine) and we will look into this today.
We can tackle the other issues after that.
Support
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mfitzgerald
Vetran
Joined: June 14 2006
Location: United States
Posts: 142
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| Posted: September 09 2008 at 1:32pm | IP Logged
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Here’s a SIP Log of the following scenario:
1. LS registers
2. Receives a call and answers.
3. LS terminates the call, our internal flag is set to
not available
4. An INVITE comes in for another call.
5. In SipIncomingCallInitialized, the code sets
IncomingCallInitEventData->TerminateIncomingCall = TRUE
and SipResponseCode = 480.
6. LS sends the 480 message for the INVITE
7. The PBX sends another INVITE for the same call
8. LS ignores it till the PBX sends the CANCEL
9. LS responds with 481…
10. Repeat 7 – 10.
This is the same as what happens when the code is
modified to correctly use AbortIncomingCall(*,0,480,
“Temporarily Unavailable”) function with one exception
(#9 never happened).
Code:
************* Log Opened (Sep 09 13:12:27) *************
>>>> TxTxTxTxTx (#1, [13:12:36.310] 0 Ms, To:
10.2.2.75:5060) >>>>
REGISTER sip:10.2.2.75 SIP/2.0
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd28117
From: <sip:7002004307@10.2.2.75>;tag=1fd2b07c
To: <sip:7002004307@10.2.2.75>
Call-Id: cda54e7a-b30d-43c1-bde7-2bd4eb6db6b2-
00000e24@10.9.1.243
CSeq: 13789811 REGISTER
Expires: 60
Max-Forwards: 70
Contact: <sip:7002004307@10.9.1.243:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#1, [13:12:36.396] 0 Ms, From:
10.2.2.75:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd28117
To: <sip:7002004307@10.2.2.75>;tag=t1220983956-co798
From: <sip:7002004307@10.2.2.75>;tag=1fd2b07c
Call-ID: cda54e7a-b30d-43c1-bde7-2bd4eb6db6b2-
00000e24@10.9.1.243
CSeq: 13789811 REGISTER
Expires: 60
Contact: <sip:7002004307@10.9.1.243:5060>
User-Agent: ENS2.2.103-IS1-RMRG0-RG3034-EP452-CO798-
CPO00000
Content-Length: 0
<<<< RxRxRxRxRx (#2, [13:12:44.830] 8434 Ms, From:
10.2.2.75:5060) <<<<
INVITE sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bc9c-33b-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220983964-co827
Date: Tue, 9 Sep 2008 18:12:44 GMT
Call-ID: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18890 INVITE
Max-Forwards: 70
X-XLData: pc=796075;dnis=2142690433
Contact: <sip:10.2.2.75:5060>
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO827-
CPO00287
Content-Type: application/sdp
Content-Length: 277
v=0
o=- 737605888 737605888 IN IP4 10.2.2.75
s=ENS Session
c=IN IP4 10.2.2.87
t=0 0
m=audio 9380 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
>>>> TxTxTxTxTx (#2, [13:12:44.876] 8566 Ms, To:
10.2.2.75:5060) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.2.2.75:5060;received=10.2.2.75;branch=z9hG4bK1a02024b
48c6bc9c-33b-1
From: <sip:10.2.2.75:5060>;tag=t1220983964-co827
To: <sip:7002004307@10.9.1.243:5060>
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Call-ID: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18890 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#3, [13:12:44.877] 1 Ms, To:
10.2.2.75:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.2.2.75:5060;received=10.2.2.75;branch=z9hG4bK1a02024b
48c6bc9c-33b-1
From: <sip:10.2.2.75:5060>;tag=t1220983964-co827
To: <sip:7002004307@10.9.1.243:5060>;tag=1fd2d1ff
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Call-ID: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18890 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#4, [13:12:44.974] 97 Ms, To:
10.2.2.75:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.2.2.75:5060;received=10.2.2.75;branch=z9hG4bK1a02024b
48c6bc9c-33b-1
From: <sip:10.2.2.75:5060>;tag=t1220983964-co827
To: <sip:7002004307@10.9.1.243>;tag=1fd2d1ff
Call-Id: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18890 INVITE
Contact: <sip:7002004307@10.9.1.243:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE,
NOTIFY
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 192
Content-Type: application/sdp
v=0
o=LanScape 2147483647 2147483647 IN IP4 10.9.1.243
s=LanScape
c=IN IP4 10.9.1.243
t=0 0
m=audio 8000 RTP/AVP 18
a=rtpmap:18 G729/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
<<<< RxRxRxRxRx (#3, [13:12:45.059] 229 Ms, From:
10.2.2.75:5060) <<<<
ACK sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bc9c-33b-2
To: <sip:7002004307@10.9.1.243:5060>;tag=1fd2d1ff
From: <sip:10.2.2.75:5060>;tag=t1220983964-co827
Call-ID: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18890 ACK
Max-Forwards: 70
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO827-
CPO00287
Content-Length: 0
>>>> TxTxTxTxTx (#5, [13:12:50.516] 5542 Ms, To:
10.2.2.75:5060) >>>>
BYE sip:10.2.2.75 SIP/2.0
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd30769
From: <sip:7002004307@10.9.1.243:5060>;tag=1fd2d1ff
To: <sip:10.2.2.75:5060>;tag=t1220983964-co827
Call-Id: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18891 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#4, [13:12:50.602] 5543 Ms, From:
10.2.2.75:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd30769
To: <sip:10.2.2.75:5060>;tag=t1220983964-co827
From: <sip:7002004307@10.9.1.243:5060>;tag=1fd2d1ff
Call-ID: 48c6bc9c0000011e00000013724ff2d1@ens.com
CSeq: 18891 BYE
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO826-
CPO00287
Content-Length: 0
<<<< RxRxRxRxRx (#5, [13:12:57.423] 6821 Ms, From:
10.2.2.75:5060) <<<<
INVITE sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bca9-364-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220983977-co868
Date: Tue, 9 Sep 2008 18:12:57 GMT
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 65790 INVITE
Max-Forwards: 70
X-XLData: pc=796080;dnis=2142690433
Contact: <sip:10.2.2.75:5060>
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO868-
CPO00298
Content-Type: application/sdp
Content-Length: 277
v=0
o=- 318597578 318597578 IN IP4 10.2.2.75
s=ENS Session
c=IN IP4 10.2.2.87
t=0 0
m=audio 9060 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
>>>> TxTxTxTxTx (#6, [13:12:57.435] 6919 Ms, To:
10.2.2.75:5060) >>>>
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP
10.2.2.75:5060;received=10.2.2.75;branch=z9hG4bK1a02024b
48c6bca9-364-1
From: <sip:10.2.2.75>;tag=t1220983977-co868
To: <sip:7002004307@10.9.1.243>;tag=f8910900
Call-Id: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 65790 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#6, [13:12:57.518] 95 Ms, From:
10.2.2.75:5060) <<<<
ACK sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bca9-364-1
To: <sip:7002004307@10.9.1.243>;tag=f8910900
From: <sip:10.2.2.75:5060>;tag=t1220983977-co868
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 65790 ACK
Max-Forwards: 70
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO868-
CPO00298
Content-Length: 0
<<<< RxRxRxRxRx (#7, [13:12:59.223] 1705 Ms, From:
10.2.2.75:5060) <<<<
INVITE sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bcab-36f-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220983979-co879
Date: Tue, 9 Sep 2008 18:12:59 GMT
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 18560 INVITE
Max-Forwards: 70
X-XLData: pc=796080;dnis=2142690433
Contact: <sip:10.2.2.75:5060>
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO879-
CPO00298
Content-Type: application/sdp
Content-Length: 277
v=0
o=- 275050855 275050855 IN IP4 10.2.2.75
s=ENS Session
c=IN IP4 10.2.2.87
t=0 0
m=audio 9060 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<<<< RxRxRxRxRx (#8, [13:13:31.343] 32120 Ms, From:
10.2.2.75:5060) <<<<
CANCEL sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bcab-36f-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220983979-co879
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 18560 CANCEL
Reason: Q.850 ;cause=102 ;text="Unknown"
Max-Forwards: 70
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO879-
CPO00298
Content-Length: 0
>>>> TxTxTxTxTx (#7, [13:13:31.361] 33926 Ms, To:
10.2.2.75:5060) >>>>
SIP/2.0 481 Transaction Does Not Exist
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bcab-36f-1
From: <sip:10.2.2.75>;tag=t1220983979-co879
To: <sip:7002004307@10.9.1.243>;tag=cf130800
Call-Id: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 18560 CANCEL
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#9, [13:13:33.223] 1880 Ms, From:
10.2.2.75:5060) <<<<
INVITE sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bccd-487-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220984013-co1159
Date: Tue, 9 Sep 2008 18:13:33 GMT
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 38780 INVITE
Max-Forwards: 70
X-XLData: pc=796080;dnis=2142690433
Contact: <sip:10.2.2.75:5060>
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO1159-
CPO00298
Content-Type: application/sdp
Content-Length: 277
v=0
o=- 891426877 891426877 IN IP4 10.2.2.75
s=ENS Session
c=IN IP4 10.2.2.87
t=0 0
m=audio 9060 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
>>>> TxTxTxTxTx (#8, [13:13:36.401] 5040 Ms, To:
10.2.2.75:5060) >>>>
REGISTER sip:10.2.2.75 SIP/2.0
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd39b4c
From: <sip:7002004307@10.2.2.75>;tag=1fd35352
To: <sip:7002004307@10.2.2.75>
Call-Id: cda54e7a-b30d-43c1-bde7-2bd4eb6db6b2-
00000e24@10.9.1.243
CSeq: 13789812 REGISTER
Expires: 60
Max-Forwards: 70
Contact: <sip:7002004307@10.9.1.243:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#10, [13:13:36.486] 3263 Ms, From:
10.2.2.75:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd39b4c
To: <sip:7002004307@10.2.2.75>;tag=t1220984016-co1175
From: <sip:7002004307@10.2.2.75>;tag=1fd35352
Call-ID: cda54e7a-b30d-43c1-bde7-2bd4eb6db6b2-
00000e24@10.9.1.243
CSeq: 13789812 REGISTER
Expires: 60
Contact: <sip:7002004307@10.9.1.243:5060>
User-Agent: ENS2.2.103-IS1-RMRG0-RG3034-EP452-CO1175-
CPO00000
Content-Length: 0
<<<< RxRxRxRxRx (#11, [13:14:04.675] 28189 Ms, From:
10.2.2.75:5060) <<<<
CANCEL sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bccd-487-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220984013-co1159
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 38780 CANCEL
Reason: Q.850 ;cause=102 ;text="Unknown"
Max-Forwards: 70
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO1159-
CPO00298
Content-Length: 0
>>>> TxTxTxTxTx (#9, [13:14:04.679] 28278 Ms, To:
10.2.2.75:5060) >>>>
SIP/2.0 481 Transaction Does Not Exist
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bccd-487-1
From: <sip:10.2.2.75>;tag=t1220984013-co1159
To: <sip:7002004307@10.9.1.243>;tag=2cc90500
Call-Id: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 38780 CANCEL
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#12, [13:14:06.224] 1549 Ms, From:
10.2.2.75:5060) <<<<
INVITE sip:7002004307@10.9.1.243:5060 SIP/2.0
Via: SIP/2.0/UDP
10.2.2.75:5060;branch=z9hG4bK1a02024b48c6bcee-4e6-1
To: <sip:7002004307@10.9.1.243:5060>
From: <sip:10.2.2.75:5060>;tag=t1220984046-co1254
Call-ID: 48c6bca90000012900000013724ff2d1@ens.com
CSeq: 43760 INVITE
Max-Forwards: 70
X-XLData: pc=796080;dnis=2142690433
Contact: <sip:10.2.2.75:5060>
User-Agent: ENS2.2.103-IS1-RMRG3134-RG3034-EP452-CO1254-
CPO00298
Content-Type: application/sdp
Content-Length: 277
v=0
o=- 265180375 265180375 IN IP4 10.2.2.75
s=ENS Session
c=IN IP4 10.2.2.87
t=0 0
m=audio 9060 RTP/AVP 18 8 0 4 101
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
>>>> TxTxTxTxTx (#10, [13:14:10.364] 5685 Ms, To:
10.2.2.75:5060) >>>>
REGISTER sip:10.2.2.75 SIP/2.0
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd43f4f
From: <sip:7002004307@10.2.2.75>;tag=1fd3f089
To: <sip:7002004307@10.2.2.75>
Call-Id: cda54e7a-b30d-43c1-bde7-2bd4eb6db6b2-
00000e24@10.9.1.243
CSeq: 13789813 REGISTER
Expires: 0
Max-Forwards: 70
Contact: <sip:7002004307@10.9.1.243:5060>;user=phone
User-Agent: LanScape VOIP Media Engine/5.12.8.9
(www.LanScapeCorp.com)
X-kgb: HV1.2.1-LS5.12.8.9-ASP1.2.0.0-CTI4.7.14070.0-
VAC4.03-PA1899
Content-Length: 0
<<<< RxRxRxRxRx (#13, [13:14:10.448] 4224 Ms, From:
10.2.2.75:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.9.1.243:5060;rport;branch=z9hG4bK1fd43f4f
To: <sip:7002004307@10.2.2.75>;tag=t1220984050-co1269
From: <sip:7002004307@10.2.2.75>;tag=1fd3f089
Call-ID: cda54e7a-b30d-43c1-bde7-2bd4eb6db6b2-
00000e24@10.9.1.243
CSeq: 13789813 REGISTER
User-Agent: ENS2.2.103-IS1-RMRG0-RG3034-EP452-CO1269-
CPO00000
Content-Length: 0
************* Log Closed (Sep 09 13:14:23) *************
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 09 2008 at 1:50pm | IP Logged
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Fitz,
Great. We will start looking right away....
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 10 2008 at 12:35pm | IP Logged
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Fitz,
Thanks for the SIP log above. SIP logs are great in helping us see what is going on at your end.
We have updated your support FTP account with “Engineering Update v5.12.8.11”. Please download and test using this updated image. Don’t forget to completely re-build your test app.
This image has been modified to better handle call processing between your VOIP app and your partivular PBX switch.
If your VOIP app now terminates an incoming call via the SipIncomingCallInitialized immedia event (or by some other means), subsequent calls your PBX is vectoring to your app should no longer be ignored by the media engine. We will have to test this at your location. We implemented the change based only on your SIP log and we did not synthesize your call flows from above.
Please test and let us know the results – especially if we messed up somewhere.
Thanks,
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 10 2008 at 12:37pm | IP Logged
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Fitz,
One other thing….
Once we know the updated version handles the “ignored calls” problem properly, we can tackle the other issues you mentioned above if required.
By the way, what IP PBX switch are you testing against?
Support
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mfitzgerald
Vetran
Joined: June 14 2006
Location: United States
Posts: 142
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| Posted: September 10 2008 at 3:45pm | IP Logged
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I have run the tests again with 5.12.8.11:
Test 1:
In TelephonyEngineCallback under the case
SipIncomingCallInitialized
Test 2:
In the thread which handles auto-answer code, within a
while loop which waits for Line State ==
SipOkToAnswerCall
Note: this thread is triggered in
TelephonyEngineCallback under the case
SipIncomingCallInitialized.
Test 3:
486 via BusyOutLine()
I was able to get this to work shortly after my previous
post. It works wonderfully. However we have not received
word on official support for 486 from the PBX provider.
FYI: Our application has built in, the ability to
check any and all libraries and dependant applications
actual version. In addition I have rechecked all
libraries, dlls and the api.h to ensure everything is up
to date.
Results:
Unfortunately the tests results remain unchanged.
Test 1:
* 480 is generated on the first INVITE received (when we
are busy)
* All subsequent INVITES for calls which have received
the 480 are ignored, even though we want to be available
* Results in a 481 from LS after a CANCEL of an ignored
INVITE.
Test 2:
* 480 is generated on the first INVITE received (when we
are busy)
* All subsequent INVITES for calls which have received
the 480 are ignored, even though we want to be available
* Does not create 481 messages
Test 3:
* Everything appears to work great. If we get word this
is officially supported we may just go this route.
To answer your question about the PBX, it is Entice.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 10 2008 at 4:37pm | IP Logged
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Ok, We will have to go further with this. We must have missed something.
We will try to actually synthesize the call flows.....
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 15 2008 at 11:52am | IP Logged
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Hi Fitz,
We are working to get you an update today. Hang tight....
Support
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 15 2008 at 4:27pm | IP Logged
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Fitz,
Please have a look at the “Engineering Update v5.12.8.12” directory in your support FTP account. Please download and test this version with your Entice IP PBX. Run your call ignore and call abort test scenarios. It should allow us to move forward.
FYI: Ask the Entice guys why they feel the need to use the same call ID in different back to back call legs. Its OK to do so as far as we see from the SIP RFC (because the INVITE request “From:” header tag is different) but if you ask us, keeping the call ID the same is asking for trouble.
This time we recreated your call flows in an attempt to resolve this issue. It never fails… we though we could simply look at your SIP call flows to fix this issue the first time but as usual, it was a bit tougher than we initially thought.
It just goes to show: Always, always, always test the software before sending it out to customers.
Here is an example call flow from another UA that calls the media engine. The media engine IGNORES the incoming call. We used the “403 Forbidden” SIP return status value (SipResponseCode = 403) when processing the SipIncomingCallInitialized immediate event.
The far end UA call the media engine again using the same call ID and the media engine answers the call.
Code:
************* Log Opened (Sep 15 16:33:11) *************
<<<< RxRxRxRxRx (#3, [16:33:22.640] 7406 Ms, From: 192.168.1.2:5060) <<<<
INVITE sip:333@192.168.1.2:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf251af1202e24d8211cb126009a533e94.0
Via: SIP/2.0/UDP 192.168.1.80:5094;rport=5094;branch=z9hG4bK356dd881;received=192.168.1.80
Record-Route: <sip:192.168.1.2:5060;lr>
From: "Extension 111" <sip:111@ps>;tag=356da20c;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7186141 INVITE
Contact: <sip:111@192.168.1.80:5094>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
max-forwards: 69
organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 246
v=0
o=111 896373166 896373166 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 16002 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTx (#3, [16:33:22.640] 7406 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf251af1202e24d8211cb126009a533e94.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356dd881
From: "Extension 111" <sip:111@ps>;tag=356da20c;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=ddc10500
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7186141 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#4, [16:33:22.656] 16 Ms, From: 192.168.1.2:5060) <<<<
ACK sip:333@192.168.1.2:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf251af1202e24d8211cb126009a533e94.0
From: "Extension 111" <sip:111@ps>;tag=356da20c;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=ddc10500
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7186141 ACK
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#5, [16:33:30.484] 7828 Ms, From: 192.168.1.2:5060) <<<<
INVITE sip:333@192.168.1.2:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf751b841f36c1d5a517f344d467c68b7c.0
Via: SIP/2.0/UDP 192.168.1.80:5094;rport=5094;branch=z9hG4bK356e255b;received=192.168.1.80
Record-Route: <sip:192.168.1.2:5060;lr>
From: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189873 INVITE
Contact: <sip:111@192.168.1.80:5094>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
max-forwards: 69
organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 246
v=0
o=111 896380997 896380997 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 16002 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTx (#4, [16:33:30.484] 7844 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf751b841f36c1d5a517f344d467c68b7c.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356e255b
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
From: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189873 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#5, [16:33:30.484] 0 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf751b841f36c1d5a517f344d467c68b7c.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356e255b
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
From: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1aeacfe
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189873 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#6, [16:33:31.265] 781 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf751b841f36c1d5a517f344d467c68b7c.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356e255b
Record-Route: <sip:192.168.1.2>
From: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1aeacfe
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189873 INVITE
Contact: <sip:333@192.168.1.2:5068>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 253
Content-Type: application/sdp
v=0
o=LanScape 2147483647 2147483647 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 20000 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRx (#6, [16:33:31.281] 797 Ms, From: 192.168.1.2:5060) <<<<
ACK sip:333@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKl751b841f36c1d5a517f344d467c68b7c
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356e255b
Route: <sip:333@192.168.1.2:5068>
From: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1aeacfe
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189873 ACK
max-forwards: 69
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#7, [16:33:32.359] 1094 Ms, To: 192.168.1.2:5060) >>>>
BYE sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01ae72be
From: <sip:333@ps>;tag=1aeacfe
To: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189874 BYE
Max-Forwards: 70
Route: <sip:192.168.1.2;lr>,<sip:111@192.168.1.80:5094>
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#7, [16:33:32.359] 1078 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5068;rport=5068;branch=z9hG4bK01ae72be;received=192.168.1.2
From: <sip:333@ps>;tag=1aeacfe
To: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189874 BYE
Proxy-Authenticate: Digest realm="ps", nonce="fc74b53e0bb6d5b430f4a7ff16088669",
opaque="8a7bf10d1b529a64e234198b68fb0b2b"
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#8, [16:33:32.359] 0 Ms, To: 192.168.1.2:5060) >>>>
BYE sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01ae845d
From: <sip:333@ps>;tag=1aeacfe
To: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189875 BYE
Max-Forwards: 70
Proxy-Authorization: Digest algorithm=md5,nonce="fc74b53e0bb6d5b430f4a7ff16088669",
opaque="8a7bf10d1b529a64e234198b68fb0b2b", realm="ps",
response="6309da62bb4a025fbf3572b4aebc3f5f",uri="sip:111@ps",username="guest"
Route: <sip:192.168.1.2;lr>,<sip:111@192.168.1.80:5094>
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#8, [16:33:32.359] 0 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5068;received=192.168.1.2;rport=5068;branch=z9hG4bK01ae845d
From: <sip:333@ps>;tag=1aeacfe
To: "Extension 111" <sip:111@ps>;tag=356e1088;x-UaId=xxxxx-yyyy-zzzzzz
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7189875 BYE
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
************* Log Closed (Sep 15 16:33:34) *************
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Here is an example call flow from another UA that calls the media engine. The media engine ABORTs the incoming call. Then the far end UA calls the media engine again using the same call ID and the media engine answers the call.
Code:
************* Log Opened (Sep 15 16:34:41) *************
<<<< RxRxRxRxRx (#15, [16:34:45.625] 20188 Ms, From: 192.168.1.2:5060) <<<<
INVITE sip:333@192.168.1.2:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf798e09b80032a4da68321181ce7ee8f9.0
Via: SIP/2.0/UDP 192.168.1.80:5094;rport=5094;branch=z9hG4bK356f3c88;received=192.168.1.80
Record-Route: <sip:192.168.1.2:5060;lr>
From: "Extension 111" <sip:111@ps>;tag=356f0302;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7267539 INVITE
Contact: <sip:111@192.168.1.80:5094>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
max-forwards: 69
organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 246
v=0
o=111 896456135 896456135 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 16002 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTx (#17, [16:34:45.625] 20188 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf798e09b80032a4da68321181ce7ee8f9.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f3c88
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
From: "Extension 111" <sip:111@ps>;tag=356f0302;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7267539 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#18, [16:34:45.625] 0 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf798e09b80032a4da68321181ce7ee8f9.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f3c88
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
From: "Extension 111" <sip:111@ps>;tag=356f0302;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1af84d2
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7267539 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#19, [16:34:47.328] 1703 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf798e09b80032a4da68321181ce7ee8f9.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f3c88
From: "Extension 111" <sip:111@ps>;tag=356f0302;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1af84d2
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7267539 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#16, [16:34:47.328] 1703 Ms, From: 192.168.1.2:5060) <<<<
ACK sip:333@192.168.1.2:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf798e09b80032a4da68321181ce7ee8f9.0
From: "Extension 111" <sip:111@ps>;tag=356f0302;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1af84d2
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7267539 ACK
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
<<<< RxRxRxRxRx (#17, [16:34:50.109] 2781 Ms, From: 192.168.1.2:5060) <<<<
INVITE sip:333@192.168.1.2:5068 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf0827edf1b7f6310ceae04b5437fd117a.0
Via: SIP/2.0/UDP 192.168.1.80:5094;rport=5094;branch=z9hG4bK356f2b04;received=192.168.1.80
Record-Route: <sip:192.168.1.2:5060;lr>
From: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294042 INVITE
Contact: <sip:111@192.168.1.80:5094>;x-inst="VGVzdCBDYWxsIERhdGEgZnJvbSB0aGUgVlBob25lIGFwcC4="
max-forwards: 69
organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 246
v=0
o=111 896460632 896460632 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 16002 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
>>>> TxTxTxTxTx (#20, [16:34:50.125] 2797 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf0827edf1b7f6310ceae04b5437fd117a.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f2b04
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
From: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294042 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#21, [16:34:50.125] 0 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf0827edf1b7f6310ceae04b5437fd117a.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f2b04
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
From: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1af9c69
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294042 INVITE
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#22, [16:34:51.312] 1187 Ms, To: 192.168.1.2:5060) >>>>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;branch=z9hG4bKf0827edf1b7f6310ceae04b5437fd117a.0
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f2b04
Record-Route: <sip:192.168.1.2>
From: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1af9c69
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294042 INVITE
Contact: <sip:333@192.168.1.2:5068>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 253
Content-Type: application/sdp
v=0
o=LanScape 2147483647 2147483647 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 20002 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=sendrecv
a=fmtp:18 annexb=no
a=ptime:20
a=fmtp:101 0-16
<<<< RxRxRxRxRx (#18, [16:34:51.328] 1219 Ms, From: 192.168.1.2:5060) <<<<
ACK sip:333@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKl0827edf1b7f6310ceae04b5437fd117a
Via: SIP/2.0/UDP 192.168.1.80:5094;received=192.168.1.80;rport=5094;branch=z9hG4bK356f2b04
Route: <sip:333@192.168.1.2:5068>
From: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
To: <sip:333@ps>;tag=1af9c69
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294042 ACK
max-forwards: 69
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#23, [16:34:53.921] 2609 Ms, To: 192.168.1.2:5060) >>>>
BYE sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01affc4a
From: <sip:333@ps>;tag=1af9c69
To: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294043 BYE
Max-Forwards: 70
Route: <sip:192.168.1.2;lr>,<sip:111@192.168.1.80:5094>
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#19, [16:34:53.921] 2593 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5068;rport=5068;branch=z9hG4bK01affc4a;received=192.168.1.2
From: <sip:333@ps>;tag=1af9c69
To: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294043 BYE
Proxy-Authenticate: Digest realm="ps",
nonce="fc74b53e0bb6d5b430f4a7ff16088669",
opaque="e81c18c9c9bd41d13c42f0f32c91e2b7"
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
>>>> TxTxTxTxTx (#24, [16:34:53.937] 16 Ms, To: 192.168.1.2:5060) >>>>
BYE sip:111@ps SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5068;rport;branch=z9hG4bK01b01e33
From: <sip:333@ps>;tag=1af9c69
To: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
Call-Id: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294044 BYE
Max-Forwards: 70
Proxy-Authorization: Digest algorithm=md5,nonce="fc74b53e0bb6d5b430f4a7ff16088669",
opaque="e81c18c9c9bd41d13c42f0f32c91e2b7", realm="ps",
response="6309da62bb4a025fbf3572b4aebc3f5f",uri="sip:111@ps",username="guest"
Route: <sip:192.168.1.2;lr>,<sip:111@192.168.1.80:5094>
User-Agent: LanScape VOIP Media Engine/5.12.8.12 (www.LanScapeCorp.com)
x-CustomHeader-Extension-333: "This is a modified transmitted SIP message."
x-PhoneLine: 0
Content-Length: 0
<<<< RxRxRxRxRx (#20, [16:34:53.937] 16 Ms, From: 192.168.1.2:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5068;received=192.168.1.2;rport=5068;branch=z9hG4bK01b01e33
From: <sip:333@ps>;tag=1af9c69
To: "Extension 111" <sip:111@ps>;tag=356f6683;x-UaId=xxxxx-yyyy-zzzzzz
Call-ID: b8254b9f-80c5-4946-8c06-9f47e7ed7bbd-00000c2c@192.168.1.80
CSeq: 7294044 BYE
x-customheader-extension-111: "This is a modified transmitted SIP message."
x-phoneline: 0
Server: LanScape Centrex Proxy/3.42.2.9 (www.LanScapeCorp.com)
Content-Length: 0
************* Log Closed (Sep 15 16:34:58) *************
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Thanks,
Support
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mfitzgerald
Vetran
Joined: June 14 2006
Location: United States
Posts: 142
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| Posted: September 17 2008 at 10:34am | IP Logged
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EUREKA… great job LS!
The 480 solution works just like it should, despite the
PBX’s use of “Re-Invite” identifiers.
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: September 17 2008 at 1:10pm | IP Logged
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Fitz,
Thanks for the feedback. Hopefully we can now move forward with other tasks.
Support
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: 480 questions
