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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 19 2008 at 1:49pm | IP Logged
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Hi Thomas,
That is not the feedback we were expecting at all. Hmmm…. Something strange is occurring. Its got to be something simple….
1)
We need to see the media engine SIP logs again for the failed calls. Please upload SIP log info to your support FTP account in a new directory and we will review. If the logs are large, that’s OK. You do not have an upload size restriction on your support FTP account.
2)
If you have some sort of log or logs from Asterisk, please upload that too. We are curious what is being reported by Asterisk.
3)
Only a guess: It is possible that there may be a simple timing issue between your VOIP Media Engine app and the SIP that is received from Asterisk. There are a bunch of API procedures we can use in the media engine to change certain call progress timeouts when setting up and tearing down SIP calls. However, default media engine configuration with Asterisk should be totally acceptable. Hmmm…. We have to think….
4)
Accessing your Asterisk box from our location: Yes, we can do this. Normally we prefer to use a simple secure VPN connection to the customer’s SIP/RTP server machine. We have not used “remote desktop” as of yet so if you want to show us how to do this, we are agreeable. What we would eventually like to do is to be able to obtain two usable extensions on your Asterisk box to use for testing. If we can access your Asterisk box from our lab via VPN connection (or globally), we can then set up two VOIP Media Engine based “call hammer test” applications we use here. We will be able to see immediately how many calls fail and pass. It would be nice if we could run a test for a few hours (or overnight) on your Asterisk server from the US.
5)
Regarding extended support: Normally we offer 2 months totally free support for the media engine. After that, we have to urge customers to take advantage of the enhanced customer support program we offer. This is paid support. We definitely want to investigate the current issue(s) you have reported. However we may have to get your company enrolled into our “enhanced customer support” program.
Respond as needed. Thanks Thomas,
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 06 2008 at 5:03am | IP Logged
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Hello Support.
Please sorry for delay, but I have two very strong weeks at hospital behind me, where was no possibility to have an internet access.
Now I have 4 or five weeks ahead to stay at home, so I can do some work before I will be back in the office.
Please be so kind and let me have an mail account where I can assist you to get full access to our machines, so you can log or delegate everything by your self and can run tests which you want. So it is possible to get access to the asterisk and also to the client.
Regards.
Thomas
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 09 2008 at 12:12pm | IP Logged
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Hi Thomas,
We can perform just about any type of remote VOIP debugging in order to assist customers. Generally customers have 2 months of free support from us after their initial purchase of the VOIP media engine. After two months have passed, customers can enter into a support contract with us so that we may fully assist them with further development, deployment or inter-op issues.
So we can assist you fully and without interruption, is it possible for you to start a support contract with us?
Thanks Thomas,
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 09 2008 at 12:34pm | IP Logged
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Hello Support,
sure is it possible to start a contract. The only one I need is, to start and simply ends, 4 calls at the same time which starts and ends well. And I think the problem is not the code or “how to do this”, the problem is the media engine which does not communicate right with an asterisk if I use vb.net. So I have to ask: are you sure that I need a support contract to find the error?
If yes please let me have a price for a code which start 4 calls, end them and start the same calls again and again….. (If you want with a simple click on a button, but without any errors)
Regards,
Thomas
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 10 2008 at 8:12am | IP Logged
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Hi Thomas,
I think the simplest approach (and quickest) would be to allow us to have access to your Asterisk box from our location. What we would do is the following:
1)
Perform call testing of your asterisk box from our location and see how the SIP call flows look. We would use LanScape call test software to perform testing. We may need a few Asterisk configured SIP extensions for testing.
2)
If we need to, code up a short example VB.NET example app that can initiate and terminate multiple calls through your asterisk box. We would pass this source code back to you for reference. This test code would be in the form of a simple VB.NET console based application.
3)
The time we spend performing steps 1 and 2 above will have to be placed under some form of paid support. We will send you additional details regarding pricing.
Summary:
Something strange must be going on for your SIP call flows not to continuously connect without errors. Hmmm… we have to think. Right now we do not have any other customer that has reported a similar issue. Sounds quite strange. Our gut tells us that it must be something simple that is causing us grief. Just to be clear, the last inter-op QA testing we performed using Asterisk was using v1.4.18.1 and we do not see the issue you report. Very strange…
Support
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 10 2008 at 10:11am | IP Logged
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Thomas,
Our CTO will contact you via email shortly with additional information on how we may move forward.
Support
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 16 2008 at 1:42pm | IP Logged
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Hi Thomas,
Based on testing that our CTO and you conducted over last week, we have created a temporary engineering release of the VOIP Media Engine you can use. Please download the “v5.12.8.4 DLL Only - Expires 12-30-08.zip” archive image from your support FTP account. The image contains release notes on changes since your version was released.
The ZIP archive contains the native code media engine DLL (LMEVoip.dll) and the .NET wrapper interface (LMEVoipManaged.dll). You can simply take these new DLL images and replace the ones you currently have.
This temporary product image is good until the end of this year. This will give us enough time to get you an official product image in the near future while still allowing you to develop and deploy your VOIP solution.
Working with you has been an enjoyable experience. We want to thank you for giving us the ability to access your equipment remotely in order to conduct proper testing. We would like you to perform initial testing with this current product image. If all looks OK, please repost to this forum thread. If all looks OK, we will email your company an invoice for the support hours that were logged.
Thanks Thomas,
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 17 2008 at 11:53am | IP Logged
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Hello Support,
there will be the same problem, but I found the reason why there will be the errors. The errors based on authentication to the asterisk. Please see my mail to Randal for more information. Now it is clear that the DLL will have the problem, net my or your code.
Hope to found a solution to this issue.
Regards
Thomas
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 18 2008 at 7:50am | IP Logged
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Do somebody read this?
regards
Thomas
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support
Administrator
Joined: January 26 2005
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Posts: 1666
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| Posted: June 18 2008 at 11:39am | IP Logged
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Hi Thomas,
Yes, we have read this post. We are a bit short-handed today with our staff. We will post as soon as we can.
Thanks,
Support
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support
Administrator
Joined: January 26 2005
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| Posted: June 19 2008 at 9:32am | IP Logged
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Hi Thomas,
Excuse the delay in this response. Yesterday was very busy here.
Randal spoke with us this morning regarding the email you sent that contained additional test information.
If your media engine phone dialer VOIP application is having authentication issues when making outgoing calls via your production asterisk boxes, then it’s a simple configuration issue in asterisk and in your VOIP Media Engine based app.
If your multi-line dialer app makes an outgoing call, Asterisk will want to authenticate the INVITE request that gets sent by the media engine to Asterisk. This will happen for every outgoing call your app (i.e. media engine) initiates.
Here is sample SIP log info from the last testing we performed Saturday using your Asterisk boxes remotely from our location:
(Note: IP addresses have been changed)
Code:
INVITE sip:1002@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5360;rport;branch=z9hG4bK00c15a0b
From: "Extension 1001" <sip:1001@asterisk>;tag=c164eb
To: <sip:1002@1.2.3.4>
Contact: <sip:1001@4.5.6.7:5360>
Call-Id: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682693 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-Source-Phone-Line: 0
Content-Length: 165
User-Agent: LanScape VOIP Media Engine/5.12.8.4 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=1001 12652218 12652218 IN IP4 4.5.6.7
s=LanScape
c=IN IP4 4.5.6.7
t=0 0
m=audio 8400 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#0, 0 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5360;branch=z9hG4bK00c15a0b;received=4.5.6.7;rport=5360
From: "Extension 1001" <sip:1001@asterisk>;tag=c164eb
To: <sip:1002@1.2.3.4>;tag=as402afc92
Call-ID: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682693 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="04e1cd91"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#1, 1 72 Ms, To: 1.2.3.4:5060) >>>>
ACK sip:1002@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5360;received=4.5.6.7;rport=5360;branch=z9hG4bK00c15a0b
From: "Extension 1001" <sip:1001@asterisk>;tag=c164eb
To: <sip:1002@1.2.3.4>;tag=as402afc92
Call-Id: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682693 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.4 (www.LanSc a peCorp.com)
x-Source-Phone-Line: 0
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#2, 0 Ms, To: 1.2.3.4:5060) >>>>
INVITE sip:1002@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5360;rport;branch=z9hG4bK00c15 a d0
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>
Contact: <sip:1001@4.5.6.7:5360>
Call-Id: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682881 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
Proxy-Authorization: Digest algorithm=md5,nonce="04e1cd91",
realm="asterisk",response="5a09f780cc49341afef82f009a395edc" ,
uri="sip:1002@1.2.3.4",username="1001"
x-Source-Phone-Line: 0
Content-Length: 165
User-Agent: LanScape VOIP Media Engine/5.12.8.4 (www.LanSc a peCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=1001 12652406 12652406 IN IP4 4.5.6.7
s=LanScape
c=IN IP4 4.5.6.7
t=0 0
m=audio 8400 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#1, 1 87 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5360;branch=z9hG4bK00c15ad0;received=4.5.6.7;rport=5360
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>
Call-ID: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1002@1.2.3.4>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#2, 1 88 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5360;branch=z9hG4bK00c15ad0;received=4.5.6.7;rport=5360
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>;tag=as0843f661
Call-ID: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1002@1.2.3.4>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#3, 6 2 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5360;branch=z9hG4bK00c15ad0;received=4.5.6.7;rport=5360
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>;tag=as0843f661
Call-ID: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682881 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1002@1.2.3.4>
Content-Type: application/sdp
Content-Length: 183
v=0
o=root 3331 3331 IN IP4 1.2.3.4
s=session
c=IN IP4 1.2.3.4
t=0 0
m=audio 8170 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#3, 4 53 Ms, To: 1.2.3.4:5060) >>>>
ACK sip:1002@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5360;received=4.5.6.7;rport=5360;branch=z9hG4bK00c15ad0
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>;tag=as0843f661
Call-Id: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682881 ACK
Max-Forwards: 70
Route: <sip:1002@1.2.3.4>
User-Agent: LanScape VOIP Media Engine/5.12.8.4 (www.LanScapeCorp.com)
x-Source-Phone-Line: 0
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#4, 4 750 Ms, To: 1.2.3.4:5060) >>>>
BYE sip:1002@1.2.3.4 SIP/2.0
Via: SIP/2.0/UDP 4.5.6.7:5360;rport;branch=z9hG4bK00c1 2 532
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>;tag=as0843f661
Call-Id: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682882 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.4 (www.LanScapeCorp.com)
x-Source-Phone-Line: 0
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#4, 4 953 Ms, From: 1.2.3.4:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.5.6.7:5360;branch=z9hG4bK00c12532;received=4.5.6.7;rport=5360
From: "Extension 1001" <sip:1001@asterisk>;tag=c165b0
To: <sip:1002@1.2.3.4>;tag=as0843f661
Call-ID: 21b6055d-6f5e-4e45-a416-f0b6c7ec0d65-0000117c@4.5.6.7
CSeq: 12682882 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1002@1.2.3.4>
Content-Length: 0
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Your multi-line dialer app (VOIP Media Engine) registered with Asterisk using its global IP address and port. The dialer app registered as extension 1001. The user name and password for extension 1001 as set in your application code were both set to “1001”. This corresponds to the values set for extension 1001 in Asterisk’s sip.conf file:
Code:
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[1001]
type=friend
secret=1001
host=dynamic
context=default
nat=yes
.
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We set your dialer app so it would always call extension 1002. Extension 1002 was terminated in our lab using the LanScape CallTester.exe application.
So if you have your voip app configured properly with the appropriate extension, amd your app makes outgoing calls, calls will not fail as the results of “407 Proxy Authentication Required” responses.
We know it works because we tested your code + media engine with your SUSE/Asterisk box in your lab.
Here is what you need to do:
1)
For the extension your voip app will use/register, make sure the sip.conf file that asterisk uses has an entry with appropriate settings. See the above portion of the config file for extension 1001 from above.
2)
Before your VOIP app registers with asterisk, make sure it calls the AddAuthorizationCredentials() API procedure to specify the user agent “user name” for your app and the proper authentication password. The user name for your app is the same as the “extension” of your app. The password for your app can be anything as long as asterisk and your app use the same password.
3)
Allow your app to register with asterisk.
4)
Allow your app to make outgoing calls. If the above REGISTER operation succeeded, then your outgoing calls will also succeed.
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 19 2008 at 12:34pm | IP Logged
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Hello Support,
1) Checked and OK
2) Aleady done
3) Now Insert
4) Checked.
Using via VPN / Internet:
100% Funktion
Using in Lokal Network 100MBit:
register success
more than 50% of ougoing calls fails
Next try, other Lan other Asterisk:
also register success
more than 50% of outgoing calls fails.
Do I cancel the user at sip.conf
and enable the guest acount that there will be no autorisation required, all calls will run fine.
Question:
Is it possible to have same code to test?
Do you want to have a local test again, because now the problem is much more detailed, and the errors can reproduced with the code that Randal send?
Any outher ideas?
Regards
Thomas
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 19 2008 at 5:53pm | IP Logged
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Thomas,
This is very strange. Hmmm…
Before we do anything more, please enable SIP logging in your VOIP application (the media engine), run call tests that show the failure and upload the SIP log to your support FTP account. We will look at it tomorrow.
We must not be testing what you are testing because last week everything we tested at your location in Germany worked fine! We tested all scenarios and did not locate a single problem. What is going on?
Its driving us a bit crazy :(
We will wait for your SIP log. That will tell us everything.
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 6:29am | IP Logged
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Hello Support,
yes the test worked fine, but after the tests I see that the guest account was open (By Randal?) and the user that Randal used (1000) was not declared at SIP.Conf. After I closed the guest account and insert user 1000 the test fails. So, now we know the reason of the failure.
The requested protokolls you ask will be the same as the first I send. What I think is that, if the event "Authorisation is requered" comes to fast, the engine will not see it. So I think it will be really a timing problem.
If you have now a need of new protokolls, please ask again.
Regards Thomas
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 20 2008 at 7:17am | IP Logged
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Thomas,
We have altered the posted Sip log as to hide your IP addresses. Our mistake. We are trying to work too fast. Thanks for pointing this out.
What do you mean when you say “the guest account was open”?
What is a “guest account”?
We are going to access your Asterisk box right now to perform a series of call tests.
Please let us know what time you are available today in case we need to speak with you.
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 7:41am | IP Logged
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Hello Support,
I am ready today for you so I cann call or mail you shortly.
Gueat Account:
If there is no Line:
allowguest=no
or a line
allowguest=yes
in the sip.conf of asterisk, there is no need to autorisation.
so you can use everything you like for user and secret of your sip properties to make a call.
And if there is no need to autorisation the event autorisation required will not fired, and everything runs fine. But if the event is fired by asterisk, the calls will run in errors.
Regards
Thomas
P.S. Please send a short mail, if I should call you
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 20 2008 at 9:02am | IP Logged
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Ok. We now understand.
We are performing a few test right now on your Asterisk box. We are accessing the asterisk box via global IP.
We will post shortly...
Thanks Thomas
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 9:14am | IP Logged
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Hello Support,
sounds good, please feel free to create your own sip account, or use the changed sip accounts which are already created.
Please think abaout that, that it will be run fine if you use the internet and the errors will be only displayed if you use a 100MBit line. (Or something else)
Regards
Thomas
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 20 2008 at 10:30am | IP Logged
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Thomas,
We finally see repeatable call failures when we access your asterisk box from our lab using your asterisk’s global IP address and port. The tests we have conducted today are exactly the test we conducted last week.
We are wondering what has changed since last week… Hmmm… Very weird.
Here is the test scenario:
1)
We remoted into your asterisk box using VNC and changed your sip.conf to include two additional extension: 1000 and 1001.
2)
We have your test code running here on a Quad core Intel host machine. We have made slight conditional compiled modification to your code. Nothing big. This version of your source code does not register to asterisk. It uses the single extension 1000. we are running it under Visual Studio 2005 IDE and debugger.
3)
We have running on the same machine the LanScape CallTester app. It is used to terminate calls coming from asterisk. It is registered as extension 1001. Auth username 1001, Auth password 1001.
4)
When we initiate outgoing call one at a time, all calls succeed as expected.
5)
When we start 30 calls all at one time, calls start to fail. The total number of calls that fail seem to be random. Note that your call dialer app (that is extension 1000) is placing 30 calls all at the same time to extension 1001 (the LanScape call tester app) via your asterisk box.
Your asterisk box somehow starts returning many, many “503 Server Error” responses. Here is the call flow that gets the errors:
Code:
Your Dialer App Asterisk
---------------------------------------
INVITE -------------->
<------------- 407 Proxy Authentication Required
ACK -------------->
INVITE -------------->
<------------- 503 Server error
ACK -------------->
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This pattern is consistent.
We are not sure why your asterisk box is sending back all these error responses. We are still looking at the individual phone line SIP logs.
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 11:10am | IP Logged
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Hello Support,
the only changes I made, is to cancel the guest account in the sip.conf.
All other situation you describe, I know these problems and agree to your result.
Randal made a lot of test on our Asterisk also from our machine to our asterisk, and say everything is OK. Also my code I send was OK.
Randal made some changes in my code (Installed on my local machine) and use there account “1000”. Randal and me wondering why his code is running. After tests of Randals code on my local machine, my code is running also.
So I ask Randal, what kind of changes he made. He said nothing, but he think it’s the new library, because I use …3 and he use …4.
But I have a try on some different asterisks and see that my and also Randals Code is only running on our test asterisk. So I check all different and the only one I see is the guest account.
Also Randal note that the server errors are more than he like, but they are ok to the equipment we use. (Some Server errors are no problem.)
I test some different apps on different servers anytime with the result that the calls fails if there is an event” autorisation required”
Make the same tests again, and cancel the line enableguest=no and all calls will running fine.
I do not know that Randal cancel this line, I use every time the account dialer and I do not know what I can do more as testing reporting and asking, because I am not in the situation to say something about the handling of events of the media engine. I test a lot of different asterisk servers in different LAN´s and I can not believe that you don’t see this errors in your LAN.
So please be so kind and let me know what I can do to use the media engine with an asterisk located in the local LAN. If there is no way to use it, I have to look for other solutions, because I spend to much money and time to this project. Feel free and see all my posts, some time ago, my name was Winkler, now it is Athineou. What a lot of problems. One more, the version I get to test, do not show this errors any time. So an old question to the support was to get an older DLL. But the answer was that there is no way to get this old versions.
An other answer from the support was: here is a new DLL which ends your entire headache. And now I have the same problems, but a bill more.
Please note that I requested all my problems any time I try to use the engine to the support, and try to find the errors at first by my self, but now I do not know what I can do, sorry
Regards
Thomas
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 20 2008 at 11:22am | IP Logged
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Thomas,
In Asterisk, is there some sort of configuration parameter that specifies the max number of calls per extension at any one time?
Support
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 11:26am | IP Logged
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Hello Support,
yes this changes are form Lanscape.
And I think from Randal.
Is there the possibility to call you?
Regards
Thomas
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 11:40am | IP Logged
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Hello Support,
I have a need to change my location and will be back in 40 min. If it is possible please be so kind a let me have a phone number where I can call you so that we can found a solution.
regards
Thomas
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BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
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| Posted: June 20 2008 at 12:23pm | IP Logged
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Hello Support,
will be back now, some news?
Regards
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: June 20 2008 at 3:39pm | IP Logged
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Thomas,
This is Randal. As we discussed in our phone call, it appears that we have been able to detect the situation you described at the beginning of this post.
Let me summarize the test we performed:
We configured an Asterisk box running Asterisk 1.14.8.1. Asterisk was set up to have 2 extensions using the SIP.CONF config file (extenstion 300 and 301). Authentication is enabled for these two extensions.
We use the LanScape “Callester” application to perform various call test scenarios at our location. We took one call tester and configured it to register with Asterisk using extension 300. We took the second call tester and configured it to register with Asterisk using extension 301. Both call tester apps were capable of initiating or receiving 64 concurrent uLaw calls.
The call testers were also set up to create individual SIP log files for each phone line. This is accomplished by calling the LogPhoneLineSipMessages() API procedure in the call tester code. Individual SIP logs for each phone line are easier to read as compared to one huge combined SIP log file.
We tried various tests that started different numbers of outgoing calls. As I said on the phone, we started testing with 16 concurrent outgoing calls and worked our way down to 4 concurrent calls where we saw the “issue” you reported. The issue being that a phone line sits in the SipStartOutgoingRing state until the call is terminated by calling the TerminateCall() API procedure.
It is normal for a phone line to remain in the SipStartOutgoingRing state for an outgoing call if no response is received from the far end of the call. Normally the media engine transmits the INVITE request for the outgoing call and then waits for responses from the call destination (Asterisk). Responses can be provisions (1XX) or any other error or authentication challenge response. Of course, responses in the 2XX range would allow the call to connect as normal.
When we finally detected a phone line that stayed in the SipStartOutgoingRing state, we terminated further testing and reviewed the phone line SIP log and the events for the line.
When a phone line sits in the SipStartOutgoingRing state for an outgoing call, this is because Asterisk has ignored the media engine transmitted INVITE or the INVITE request was not delivered to Asterisk because of a network error.
Here are the events that were logged for the phone line that stayed in the SipStartOutgoingRing state:
Code:
1)
This is the last good outgoing call made to Asterisk.
SipOutgoingCallInitializing
SipOutgoingCallStart
SipDialTone
SipDialing
SipSendInvite
SipModifySipMessage
SipStartOutgoingRing
SipModifySipMessage
SipModifySipMessage
SipReceivedProvisionalResponse
SipReceived100Trying
SipWaitForInviteOk
SipReceived180Ringing
SipInviteOkReceived
SipSendInviteAck
SipModifySipMessage
SipOutgoingCallConnected
SipInCall
SipSendBye
SipModifySipMessage
SipReceivedByeAck
SipCallComplete
SipOnHook
2)
On this next outgoing call to Asterisk, we receive a "503 Server Error".
We are not sure why Astersisk returned the 503 response. SIP for the INVITE
request looked good.
SipOutgoingCallInitializing
SipOutgoingCallStart
SipDialTone
SipDialing
SipSendInvite
SipModifySipMessage
SipStartOutgoingRing
SipModifySipMessage
SipModifySipMessage
SipFarEndError
SipModifySipMessage
SipOnHook
3)
Here is where we start the final outgoing call that gets no response to
the transmitted INVITE that contains authentication credentials. Just
like your testing, the phone line remains in the "SipStartOutgoingRing"
state. This is because Asterisk is appearing to ignore the call's
transmitted INVITE request the media engine sent.
SipOutgoingCallInitializing
SipOutgoingCallStart
SipDialTone
SipDialing
SipSendInvite
SipModifySipMessage
SipStartOutgoingRing
SipModifySipMessage
SipModifySipMessage
SipModifySipMessage
4)
Here is where the call gets terminated.
SipCallCanceled
SipOnHook
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Also, here is the SIP log for the phone line:
Code:
The last good call:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#65, [14:17:08.125] 657 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2a619
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2711a
To: <sip:301@192.168.1.122>
Contact: <sip:300@192.168.1.2:5060>
Call-Id: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709573 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-Source-Phone-Line: 1
Content-Length: 162
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 28469046 28469046 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8054 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#65, [14:17:08.140] 672 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2a619;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2711a
To: <sip:301@192.168.1.122>;tag=as5ff19063
Call-ID: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709573 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e6632ee"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#66, [14:17:08.140] 15 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;rport=5060;branch=z9hG4bK01b2a619
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2711a
To: <sip:301@192.168.1.122>;tag=as5ff19063
Call-Id: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709573 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#67, [14:17:08.218] 78 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2a67a
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>
Contact: <sip:300@192.168.1.2:5060>
Call-Id: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709666 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
Proxy-Authorization: Digest algorithm=md5,nonce="0e6632ee",realm="asterisk",
response="2a1ab9662bcf3ad43d5b74cb58431c66",uri="sip:301@192.168.1.122",username="300"
x-Source-Phone-Line: 1
Content-Length: 162
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 28469125 28469125 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8054 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#66, [14:17:08.265] 125 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2a67a;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>
Call-ID: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.122>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#67, [14:17:08.265] 0 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2a67a;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>;tag=as6886e1c4
Call-ID: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.122>
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#68, [14:17:08.296] 31 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2a67a;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>;tag=as6886e1c4
Call-ID: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709666 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.122>
Content-Type: application/sdp
Content-Length: 184
v=0
o=root 2325 2325 IN IP4 192.168.1.122
s=session
c=IN IP4 192.168.1.122
t=0 0
m=audio 17748 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#68, [14:17:08.296] 78 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;rport=5060;branch=z9hG4bK01b2a67a
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>;tag=as6886e1c4
Call-Id: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709666 ACK
Max-Forwards: 70
Route: <sip:301@192.168.1.122>
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#69, [14:17:09.437] 1141 Ms, To: 192.168.1.122:5060) >>>>
BYE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2945f
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>;tag=as6886e1c4
Call-Id: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709667 BYE
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#69, [14:17:09.437] 1141 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2945f;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2717b
To: <sip:301@192.168.1.122>;tag=as6886e1c4
Call-ID: 41bf3a15-c7f1-4182-b161-c0c1aa96b775-00000bf8@192.168.1.2
CSeq: 11709667 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.122>
Content-Length: 0
The call that received the "503 server error" from Asterisk:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#70, [14:17:09.984] 547 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2736f
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2a2a1
To: <sip:301@192.168.1.122>
Contact: <sip:300@192.168.1.2:5060>
Call-Id: 15bc8a4b-5486-40c5-839d-3a12bac48b2d-00000bf8@192.168.1.2
CSeq: 11702743 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-Source-Phone-Line: 1
Content-Length: 162
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 28470953 28470953 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#70, [14:17:10.000] 563 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2736f;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2a2a1
To: <sip:301@192.168.1.122>;tag=as67b63155
Call-ID: 15bc8a4b-5486-40c5-839d-3a12bac48b2d-00000bf8@192.168.1.2
CSeq: 11702743 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61810bb0"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#71, [14:17:10.015] 31 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;rport=5060;branch=z9hG4bK01b2736f
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2a2a1
To: <sip:301@192.168.1.122>;tag=as67b63155
Call-Id: 15bc8a4b-5486-40c5-839d-3a12bac48b2d-00000bf8@192.168.1.2
CSeq: 11702743 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#72, [14:17:10.031] 16 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2b968
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2ccce
To: <sip:301@192.168.1.122>
Contact: <sip:300@192.168.1.2:5060>
Call-Id: 15bc8a4b-5486-40c5-839d-3a12bac48b2d-00000bf8@192.168.1.2
CSeq: 11694100 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
Proxy-Authorization: Digest algorithm=md5,nonce="61810bb0",realm="asterisk",
response="7ff7d3b9f20d1abe245c03244663f9e0",uri="sip:301@192.168.1.122",username="300"
x-Source-Phone-Line: 1
Content-Length: 162
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 28471000 28471000 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8000 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#71, [14:17:10.062] 62 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2b968;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2ccce
To: <sip:301@192.168.1.122>;tag=as67b63155
Call-ID: 15bc8a4b-5486-40c5-839d-3a12bac48b2d-00000bf8@192.168.1.2
CSeq: 11694100 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.122>
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#73, [14:17:10.093] 62 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;rport=5060;branch=z9hG4bK01b2b968
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2ccce
To: <sip:301@192.168.1.122>;tag=as67b63155
Call-Id: 15bc8a4b-5486-40c5-839d-3a12bac48b2d-00000bf8@192.168.1.2
CSeq: 11694100 ACK
Max-Forwards: 70
Route: <sip:301@192.168.1.122>
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
The call that received no further responses after sending a second INVITE
to Asterisk with authentication credentials:
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#74, [14:17:10.281] 188 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2ba66
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2cdcc
To: <sip:301@192.168.1.122>
Contact: <sip:300@192.168.1.2:5060>
Call-Id: 4bada667-bd66-4e7e-8c7f-9f0e6f9e52b2-00000bf8@192.168.1.2
CSeq: 11694350 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
x-Source-Phone-Line: 1
Content-Length: 162
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 28471250 28471250 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8054 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
<<<< RxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (#72, [14:17:10.281] 219 Ms, From: 192.168.1.122:5060) <<<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK01b2ba66;received=192.168.1.2;rport=5060
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2cdcc
To: <sip:301@192.168.1.122>;tag=as152a2e0d
Call-ID: 4bada667-bd66-4e7e-8c7f-9f0e6f9e52b2-00000bf8@192.168.1.2
CSeq: 11694350 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="21f80d54"
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#75, [14:17:10.281] 0 Ms, To: 192.168.1.122:5060) >>>>
ACK sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;received=192.168.1.2;rport=5060;branch=z9hG4bK01b2ba66
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2cdcc
To: <sip:301@192.168.1.122>;tag=as152a2e0d
Call-Id: 4bada667-bd66-4e7e-8c7f-9f0e6f9e52b2-00000bf8@192.168.1.2
CSeq: 11694350 ACK
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#76, [14:17:10.296] 15 Ms, To: 192.168.1.122:5060) >>>>
INVITE sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2ba76
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2cddc
To: <sip:301@192.168.1.122>
Contact: <sip:300@192.168.1.2:5060>
Call-Id: 4bada667-bd66-4e7e-8c7f-9f0e6f9e52b2-00000bf8@192.168.1.2
CSeq: 11694350 INVITE
Max-Forwards: 70
Organization: 44388BAF-8E86-4E68-8CB3-4ED9BB21E9C4
Proxy-Authorization: Digest algorithm=md5,nonce="21f80d54",realm="asterisk",
response="d7800978f29caede9636839be2b76b24",uri="sip:301@192.168.1.122",username="300"
x-Source-Phone-Line: 1
Content-Length: 162
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
v=0
o=300 28471265 28471265 IN IP4 192.168.1.2
s=LanScape
c=IN IP4 192.168.1.2
t=0 0
m=audio 8054 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv
a=ptime:20
At this point, the phone line state sould be SipStartOutgoingRing (ignoring
all SipModifySipMessage events)
This CANCEL is then sent out whenthe call gets terminated by the CallTester.
No response wes received for this. We are looking into why this may be.
>>>> TxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (#77, [14:17:17.718] 7422 Ms, To: 192.168.1.122:5060) >>>>
CANCEL sip:301@192.168.1.122 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;rport;branch=z9hG4bK01b2ba76
From: "Extension 300" <sip:300@asterisk122.lslab.com>;tag=1b2cddc
To: <sip:301@192.168.1.122>
Call-Id: 4bada667-bd66-4e7e-8c7f-9f0e6f9e52b2-00000bf8@192.168.1.2
CSeq: 11694350 CANCEL
Max-Forwards: 70
User-Agent: LanScape VOIP Media Engine/5.12.8.5 (www.LanScapeCorp.com)
x-Source-Phone-Line: 1
Content-Length: 0
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I am assuming that what we reproduced here is the same issue you are seeing at your location with your Asterisk servers.
I wanted to post this test info so that you can verify and tell us if your situation is the same.
We will be performing additional test to make sure the final call INVITE and CANCEL was transmitted by the media engine. We’ll use WireShark to capture the data to verify. Its possible there may be a bug in the media engine that caused the final call INVITE and CANCEL to not be transmitted but this is only a guess.
We will post more info as we test further. Let me know if I am completely of base or this is not the issue you are seeing.
Randal
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Lost Calls .Net
