| Author |
|
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: April 25 2008 at 2:11pm | IP Logged
|
|
|
Hy There
Now I have a lot of time to test the voice media engine with visual studio 2005 and 2008 both on XP and Vista with 32 and 64 bit on slowly, normal and really high end machines and have a lot of problems.
First, I ask it before, I do not can make more as one call at a time if I start the engine with the parameter linemode is in SWITCH_LINE. Your message before and the Helpfile say that it must be set to switchline, if I want to make more the one Call at a time but it is only possible to make more then one call if the parameter is in PhoneLine mode. Please check the example you send to make two calls there is also the parameter set to phone line mode. Please advise me which one is right, the example you send or the Helpfile and your notice to this topic.
Second problem I have is to make much more calls at a time.
First I wondering that I make exactly 100 Calls with a maximum of 10 calls a time and the PBX made only 78 outgoing calls and 22 times the PBX log an error that the PBX tried to find the client, but no response.
It seems that the communication between the PBX and the media engine is not perfect.
After this I have a try with the same code running on the same machine but use a very very fast PBX. The result of this test where 100 calls try go out, the PBX returns 9 really calls and 91 errors with no communication to the client.
After this one, I try the same code on a very high end, very fast machine and a normal working PBX with the same result.
After this I have a try with the same code on a normal machine and a normal working PBX but the communication between the PBX and the media engine is a very slow VPN connected over an ADSL Internet connection, slower than 1 MBit and see … it works.
With this configuration a make 1000 outgoing Calls with a maximum of 10 calls a time, (No call will be answered, only ringing) and the PBX note 985 outgoing calls and 15 Server Errors.
These 15 Errors I get the information wit the event SipModifySipMessage before SipFarEndIsBusy but I do not found a solution to get the responding phone line. If there will be a solution to get this information I would be very happy to every hint.
So, I make changes in my code to save more and more time especially to the MediaEngineCallbackProc with the result that if the software will be go faster and faster the errors will be more and more.
It seems, if the communication between PBX and media engine is very slow, every thing is working, if there comes the information from the PBX to the media engine to fast, the media engine will lost some information. The media engine is not the first telephone engine I test. The most engines before have a problem with the communication between managed and unmanaged code. And it seems the media engine has the same problem.
With the trying version of the media engine there where no problem to make some calls at a time, and I do not remember me that there was a problem if I am working on a high end machine, so if there will be the possibility to get these old version for my licence, why not an older version if them works?
So please be so kind and let me know if there will be a known problem and if there is a possibility to get some more information or working solutions with the problem of the linemode parameter. If you have a need of some more information, log files ore events which the media engine sends before the communication to the PBX will be lost, please advice.
Last question I have is about the Development Resources topic of your homepage. There are 13 files with more information and help and no one is available. If there is the possibility to get more information or help in form of white papers, examples ore code snipes, specially to .Net code, please let me know.
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: April 28 2008 at 11:53am | IP Logged
|
|
|
Hi Thomas,
Thanks for posting. We have a "shut down" day today for maintenance. We will be able to respond to your post right away tomorrow morning.
Thanks,
Support
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: April 29 2008 at 12:07pm | IP Logged
|
|
|
Hi Thomas,
Good to receive your post again. VERY IMPORTANT - We need to get you an update to the most current product image we offer. From our information, you should have v5.12.7.22 - 32 line media engine. Is this correct?
Also, we may have a slight language barrier between us so if we post information for you and you don’t understand, please ask us to clarify as needed. We will do the same if required.
PHONE_LINE and SWITCH_LINE modes:
PHONE_LINE mode is not supported anymore so completely forget that it ever existed. If you have a version of the media engine that supported PHONE_LINE mode, don’t use it. We will need to get you updated software.
If you place the media engine into SWITCH_LINE mode, you will be able to start many calls at the same time. If you can’t start many calls at the same time, something else must be causing the problem.
About your test results:
If you are running v5.12.7.22, the current version will remove a host of issues that may be causing you pain. Based on your good descriptions, it looks like you have run many excellent tests. We need you to start testing with the latest product image (v5.12.8.1). Let’s get you updated and then move forward from here.
Development Resources topic of your homepage:
Yes, we have been very bad in getting the additional developer documents completed that are on the dev resource web page. All we can say is: Until we produce that content, all of your development questions will have to be answered via the support forum. We will help any way we can. We offer 2 months free support from the date of purchase for the media engine and after that we offer paid support. You also have an option to dump the free support and enter into a paid support agreement with us immediately. That way we will update your software whenever new versions are available and we can get you whatever C/C++ or .NET code samples you may require.
The absolute first thing we need to do before moving forward is to get your software updated immediately.
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: April 29 2008 at 1:49pm | IP Logged
|
|
|
Yes you are right, the version I use will be v5.12.7.22 - 32 line .
Please let me know how to update the engine and I will start my tests again.
Please sorry about my English......
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: April 29 2008 at 2:19pm | IP Logged
|
|
|
Hi Thomas,
We will generate a new product image for you to download via your support FTP account. You will be upgraded to v5.12.8.1. We will repost when your product image is ready to download. You should already have your support FTP login information from your original order confirmation you received from us on December 10, 2007.
Thank you Thomas,
Support
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: April 29 2008 at 3:29pm | IP Logged
|
|
|
Hi Thomas,
Your new media engine product image is ready for download. Please see the v5.12.8.1 directory in your support FTP account.
Thanks,
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: April 29 2008 at 3:38pm | IP Logged
|
|
|
Thank you for the update,
the download is running, the results off my tests will be need any more time.
So I will post again...
Ragards and greetings from Germany
Thomas
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: April 30 2008 at 12:22pm | IP Logged
|
|
|
Hello support,
After installing the new version of the media engine, I test my project.
First I change the code to use the switch line mode. Now it is possible to make more than on call at a time in switch line mode, no other changes where needed.
This point of trouble is solved, thanks a lot.
The problem with timing is not solved but now I get some more events in case of failure.
To be sure to use only the new version of the media engine, I create a new project. Most of the code will be copied from the “SimpleMakeCall” sample.
This code can only make a new call ( with a click on a button starts a new call, so two clicks two calls etc), hang up a call and displays the events.
First, I check und use the code from my home office in debug mode, after that with a setup and installed version of this test. Both tests are successfully works fine. The home office is connected via 0.8Mbit DSL VPN to the PBX.
Both tests are made again in my normal office where the PBX is connected with a 100Mbit connection. The used machine to run the code is an Intel Quad Core CPU 2,4Ghz 8GByte Ram on Vista 64Bit. (And some other nice parts like a very fast Raid10 Hard drive …I like this one….) The results of this test are that the most of the calls will failed. (More than 80 of a 100) Because it is not a lot off work, I have some more tried’s with some other computers and it looks like if, is the computer slower, there is less error.
So we have an old one with Windows XP, less memory. On this machine, much more calls will work. (Maybe 30 of a 100 will failed)
So a call, which is going right, sends these events:
19 SipModifySipMessage
18 SipFarEndIsBusy
17 SipModifySipMessage
16 SipReceived183SessionProgress
15 SipModifySipMessage
14 SipWaitForInviteOk
13 SipReceived100Trying
12 SipReceivedProvisionalResponse
11 SipModifySipMessage
10 SipModifySipMessage
9 SipModifySipMessage
8 SipModifySipMessage
7 SipStartOutgoingRing
6 SipModifySipMessage
5 SipSendInvite
4 SipDialing
3 SipDialTone
2 SipOutgoingCallStart
1 SipOutgoingCallInitializing
(The end is busy, its ok, if connect the corresponding events will be send)
But the most of the calls will be stopped at step 10:
10 SipModifySipMessage
9 SipModifySipMessage
8 SipModifySipMessage
7 SipStartOutgoingRing
6 SipModifySipMessage
5 SipSendInvite
4 SipDialing
3 SipDialTone
2 SipOutgoingCallStart
1 SipOutgoingCallInitializing
In case of no new command will be send (terminate ...) there will be no more events.
In case of terminate the call there come some more events:
21 SipModifySipMessage
20 SipModifySipMessage
19 SipModifySipMessage
18 SipModifySipMessage
17 SipOnHook
16 SipCallCanceled
15 SipModifySipMessage
14 SipModifySipMessage
13 SipModifySipMessage
12 SipModifySipMessage
11 SipModifySipMessage
The events 11 to 14 comes shortly after send the terminate command, the other events comes every 1 or two seconds.
The Sip message number 13, 14, 15,18,19,20 and 21will be always like:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.27.6:5060;branch=z9hG4bK6a68880c;received=192.168.27 .6;rport=5060
From: "Extension dialer" <sip:dialer@192.168.27.5>;tag=6a681937
To: <sip:03099999@192.168.27.5>;tag=as68811c61
Call-ID: 3b1aa736-235a-463e-897b-fab326cca4f5-00001038@192.168.27.6
CSeq: 6836530 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
If the “terminate call” command comes very late, the PBX will terminate this call, note an error like “Client not answer” and the sip message will be like “no existing call”
The PBX sends in both cases the error.
At least, I made the first test again from my home office via DSL and VPN and all calls will be work.
Any idea what’s happen?
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: April 30 2008 at 4:08pm | IP Logged
|
|
|
Hi Thomas,
A fast host computer should not cause any call connect problems. Unfortunately, something strange must be going on.
From your description, it looks like Asterisk is not seeing the INVITE request that the test app is sending. Have you looked at the Asterisk SIP log to see if all INVITES are being received by Asterisk?
An event sequence like the one you described:
Code:
10 SipModifySipMessage
9 SipModifySipMessage
8 SipModifySipMessage
7 SipStartOutgoingRing
6 SipModifySipMessage
5 SipSendInvite
4 SipDialing
3 SipDialTone
2 SipOutgoingCallStart
1 SipOutgoingCallInitializing
|
|
|
indicates that the media engine successfully transmitted the INVITE request for the call but received no response at all from your Asterisk box. Hmmm… strange.
Another alternative is to use the open source WireShark or Ethereal network sniffers to capture all the SIP INVITEs that are being sent to your asterisk box. The media engine should be transmitting all the INVITES properly. If not, then there is an issue inside the media engine.
The only real way for us to completely see what is going on between your test application running on the Quad Core CPU and the Asterisk server would be to look at a media engine generate SIP log. We need to make sure that INVITE requests are being sent to the proper SIP IP:port of your asterisk box for all calls.
Test Plan:
Please run your tests again with the Quad Core computer with SIP logging enabled to a file. To turn ON SIP file logging in the media engine, set appropriate values for:
BOOL LogSipMessages;
char *pSipLogFileName;
in the START_SIP_TELEPHONY_PARAMS structure that get passed to the StartSipTelephony() API procedure. Please see the Software Developer’s Reference for additional information.
Your support FTP account is enabled so please upload your SIP log there. We will review the SIP log as soon as you upload it. If the SIP log is long, that's OK.
Thanks,
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: April 30 2008 at 11:15pm | IP Logged
|
|
|
Hello Support
the ftp account is enabled but no permission to create new files.
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 01 2008 at 7:20am | IP Logged
|
|
|
Oops...
It should be OK now.
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: May 01 2008 at 1:06pm | IP Logged
|
|
|
Hello,
I upload 3 files to the ftp, one with a call that will be ok, the other one with a call which fails and a larger file with a lot of calls.
Hopefully there is some help inside.
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 01 2008 at 1:45pm | IP Logged
|
|
|
Thanks Thomas, we will review....
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 05 2008 at 1:39pm | IP Logged
|
|
|
Thomas,
Question 1:
What version and distribution of Asterisk are you using?
Support
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 05 2008 at 2:08pm | IP Logged
|
|
|
Thomas,
We think that it may be related to this post. Probably something in Asterisk has changed since we tested last. Your SIP log shows similar behavior.
SIP 487 Request Terminated:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=477&PN =1&TPN=1
Support
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 06 2008 at 10:58am | IP Logged
|
|
|
Thomas,
What version and distribution of Asterisk are you using?
Support
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 06 2008 at 3:39pm | IP Logged
|
|
|
Hi Thomas,
The calls were not actually failing. It appears to be a timing issue associated with canceling media engine outgoing calls and asterisk’s response times.
The media engine is waiting a fixed amount of time for the call's CANCEL requests to be received. Internally the media engine uses a value of 2000 Ms to wait for CANCEL related responses back from a server. If the media engine doesn't get the responses, then the call is terminated and we tear down the call. Asterisk in the meantime will timeout its phone line (and retransmit 487 Transaction terminated) because we will never send him the final INVITE ACK.
We have updated the media engine to allow app software to change the call cancel timeout used. It will be available in the next release.
You can also look at this post. Jalal recently reported a similar finding:
SIP 487 Request Terminated:
http://www.lanscapecorp.com/forum/forum_posts.asp?TID=477&PN =1
In the mean time until our next release, see if you can configure your asterisk boxes to lower the call cancel timeout, then this issue can be minimized and your asterisk phone lines will not be unavailable for longer than necessary.
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: May 07 2008 at 1:57pm | IP Logged
|
|
|
Hello,
Please sorry for delay, I have to stay a couple of days at the hospital.
I use some different versions of Asterisk, first an older 1.0.9 Bristuffed on an red hat Linux,
But the most tests I run on 1.4.17 Bristuffed 040.test 6 where running a suse Linux 10.3 kernel 2.6.22.13
But we have the same problems with Asterisk 1.2.26 bristuffed.
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 07 2008 at 2:50pm | IP Logged
|
|
|
Hi Thomas,
Hospital? We hope you are ok and it is nothing serious.
Thanks for your additional test info. Yes, there is something going on. We will be looking at it this week and get it resolved as soon as we can. We think its probably something simple. We will repost when we find the problem and solution.
Don’t worry at all. We will get you a new free product image with the fixes. :)
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: May 08 2008 at 4:34am | IP Logged
|
|
|
Hello,
it sounds good, that you found an error and I can get an update.
So I have to stay some more weeks at the hospital, but the most time it will be able to write some code to test a new version. (Thanks to UltraVNC and remote desktop…) Also there will be the possibility to read the forum, but please be sorry if there is a small delay in my answers.
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 08 2008 at 10:51am | IP Logged
|
|
|
Hi Thomas,
Yup, VNC is great.
We hope for your very speedy recovery. :(
Everyone here sends you are best wishes. Get well soon.
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: May 14 2008 at 1:29am | IP Logged
|
|
|
Hello Support,
Thank you for your wishes.
I would ask if you have some news or an estimated time for our problem.
Regards
Thomas
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 14 2008 at 7:00am | IP Logged
|
|
|
Hi Thomas,
We have been working on this issue this week. We should be able to have the needed changes implemented fairly soon. Hopefully this week.
Support
|
| Back to Top |
|
| |
support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
|
| Posted: May 16 2008 at 5:45pm | IP Logged
|
|
|
Hi Thomas,
We hope you are well.
We have an updated product image you can use temporarily for your VOIP development and testing. The call failure(s) that you reported with Asterisk should be fixed.
This temp image is actually a “sneak preview” of the v6 offering soon to be released. You can use this ‘DLL only” product image with your current installation by simply replacing your existing files with the new ones in the image.
You can download the temp development image from your support FTP account. See the “DLL Only v5.12.8.3” directory of your support FTP account. Note that this distribution only contains minimal content so that you can preview the new changes and continue with your development.
Please read the release notes that accompany the product image for the latest changes and news.
If you want to enable new v6 integrated DTMF related functionality, you will have to use the temp license that comes with the image. The trial license is good for a couple of months. If you eventually want to enable the new integrated DTMF capabilities (fully integrated in-band and RFC2833 DTMF generation and detection) you will have to purchase an upgrade to v6 sometime later this year. Otherwise we will be able to issue you a new v5 product license at no cost. In this case the v5 product license will not have internal DTMF capabilities enabled.
Repost with any questions feedback.
Thanks Thomas,
Support
|
| Back to Top |
|
| |
BMV_Thomas
Intermediate
Joined: December 26 2007
Location: Germany
Posts: 32
|
| Posted: May 17 2008 at 5:02pm | IP Logged
|
|
|
Hello Support.
Thanks for wishes but I have to stay some more weeks at hospital.
But anyway, I thank you for update, but there is not really something better.
Asterisk log the same warnings and most of the outgoing calls fails.
Now there are more calls which going right. 1/3 working 2/3 stops with errors, but
this is really not, what I am looking for.
So what I can offer is a remote desktop to the vista machine and a SSH login to the asterisk if it will help. Based of my situation I do not make any other tests but it seems the base problem is not solved.
Regards
Thomas
|
| Back to Top |
|
| |
Active Topics 
Member List 
Search 
Help 
Register 
Login
LanScape Support Forum -> LanScape VOIP Media Engine™ - Technical Support
Topic: Lost Calls .Net
