Problem with Linphone call test

[Desi Pepova]

Hello guys!

                  I have troubles making test call with Linphone. Everytime I tries it says: No answer. What may be the problem?

[George Vagenas]

Hi Desi,

I remember we troubleshoot together the problem you had to connect to the Restcomm AMI, so now I guess you can connect and check logs if needed.

For the problem you have can you please provide more information? Such as 

• What number are you trying to dial with Linphone?
• Can you try to dial using another softphone? For example Jitsi https://jitsi.org/
• What is the message you see on the Linphone?

Thanks

George

On Mon, Oct 28, 2013 at 11:20 PM, Desi Pepova <miss...@gmail.com> wrote:

Hello guys!

                  I have troubles making test call with Linphone. Everytime I tries it says: No answer. What may be the problem?

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George Vagenas

[Desi Pepova]

The number I am dialing is 1234, as it is shown in the example, and it says: No answer. And when I tried with Jitsi it said  Failed to create the call.

[Desi Pepova]

I havent't got elasti IP adress, only private DNS.

On Monday, October 28, 2013 11:20:20 PM UTC+2, Desi Pepova wrote:

[Desi Pepova]

I already run the script, Public DNS: ec2-54-237-100-50.compute-1.amazonaws.com.

On Monday, October 28, 2013 11:20:20 PM UTC+2, Desi Pepova wrote:

[George Vagenas]

Hi Desi,

Please try with Jitsi sip client. I did a successful call to the hello-play application using this SIP URI : sip:12...@ec2-54-237-100-50.compute-1.amazonaws.com:5080

George

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George Vagenas

[Desi Pepova]

When I try to make a call in Jitsi the following error occurs: "Failed to create the call. Invalid transport." And when I click to see more details the following is shown:

java.lang.IllegalArgumentException: invalid transport

at net.java.sip.communicator.impl.protocol.sip.SipStackSharing.getJainSipProvider(SipStackSharing.java:428)

at net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getJainSipProvider(ProtocolProviderServiceSipImpl.java:1465)

at net.java.sip.communicator.impl.protocol.sip.ProtocolProviderServiceSipImpl.getDefaultJainSipProvider(ProtocolProviderServiceSipImpl.java:1829)

at net.java.sip.communicator.impl.protocol.sip.SipMessageFactory.createInviteRequest(SipMessageFactory.java:723)

at net.java.sip.communicator.impl.protocol.sip.SipMessageFactory.createInviteRequest(SipMessageFactory.java:828)

at net.java.sip.communicator.impl.protocol.sip.CallSipImpl.invite(CallSipImpl.java:347)

at net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.createOutgoingCall(OperationSetBasicTelephonySipImpl.java:174)

at net.java.sip.communicator.impl.protocol.sip.OperationSetBasicTelephonySipImpl.createCall(OperationSetBasicTelephonySipImpl.java:119)

at net.java.sip.communicator.service.protocol.media.AbstractOperationSetBasicTelephony.createCall(AbstractOperationSetBasicTelephony.java:110)

at net.java.sip.communicator.impl.gui.main.call.CallManager.call(CallManager.java:1895)

at net.java.sip.communicator.impl.gui.main.call.CallManager.access$800(CallManager.java:46)

at net.java.sip.communicator.impl.gui.main.call.CallManager$CreateCallThread.run(CallManager.java:1793)

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Desislava Hincheva

mobile: 0895 61 41 73

[George Vagenas]

Hi Desi,

Something is wrong with the configuration of the softphone. With Jitsi you need to create a Registraless account with no proxy or registration server. 

George

[Desi Pepova]

How I can do that?

On Monday, October 28, 2013 11:20:20 PM UTC+2, Desi Pepova wrote:

[George Vagenas]

Check Jitsi documentation here: https://jitsi.org/Documentation/RegistrarlessSIPAccount

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George Vagenas

[Desi Pepova]

I made another registration and It says: "The remote party has not replied. The call will be disconnected."

On Monday, October 28, 2013 11:20:20 PM UTC+2, Desi Pepova wrote:

[la3...@gmail.com]

Hi

I don't know.  I have had exactly the same kinds of problems describedin this thread.   However, I did some analysis using Wireshark and this is what I saw when dialing using linphone.

The first column is the last two digits of the IP address of the server running Restcomm.  What's important in this dump I think is that there are no packets coming to my workstation running linphone.  I'm guessing that there are some firewall issues in my setup. 

When using jitsi I get an assortment of error messages (most of them mentioned elsewhere in this forum), but no connection. However, of the ten or so attempts I made two of them actually made it into Restcomm's logs:

No ringing was heard, and the jitsi client ended up giving some error message that I now don't remember.

One thing I would have liked to do was to run a tcpdump from the inside of the appliance to see if there are any packets arriving at it, and even more importantly, being sent from it.   This kind of observation would be able to confirm directly my suspicion that there is a firewall issue, and also help me fix that issue right away.

Is there any way I can log into the appliance and run a little tpcdump there?

Best wishes

Bjørn

[la3...@gmail.com]

Hi.  I've been trying to get a SIP client to talk to the appliance a while now, and I have had no luck so far. I have tried both jitsi and linfone.

With linfone I set up tcpdump to look at the traffic, this is what I found as seen through Wireshark:

The first column of the image above are the last two digits of the IP number associated with the appliance.  Now the thing to notice here is that there are -no- packets coming -from- to appliance to my workstation.

I tried using jitsi, and I got an assortement of error messages and java stracktraces, but I also got two (but no more!) log entries in the restcomm appliance:

Put together this evidence seems to indicate that at last some traffic arrives at the restcomm appliance. Since no SIP traffic arrives from the appliance at my workstation, I am guessing that there is some firewall issue.  At this point it would have been extremely useful to set up a wireshark dump on the appliance to see if it sends any packets  towards my workstation.   I've tried logging into it using ssh, but so far I have had no luck doing that. Is ssh into the appliance supposed to work or not?

I'll continue exploring this tomorrow, but if anyone have any hints I can use before then I'm only too happy to try them ;-)

[Bjørn Remseth]

[George Vagenas]

Hi Bjørn,

You can login to your EC2 instance using the following command for Linux:

  ssh -i you_pem_file.pem customer@EC2_PUBLIC_IP_ADDRESS

In case you are using Windows check Amazon instructions for Putty but keep in mind that username will be customer : http://docs.aws.amazon.com/AWSEC2/latest/UserGuide/putty.html

Also, check the Restcomm AMI quick start guide here: http://www.telestax.com/restcomm-ami-quick-start-guide/

Now to your problem maybe Media server is not running or is not configured correct. 

One case that Restcomm instance is not configured correct, is when you first create the EC2 instance and then you assign to it an Elastic IP address. For this case you need to restart the instance for the auto-configuration scripts to pick up the Elastic IP Address and setup Restcomm properly.

After you ssh to the instance you can collect and send the logs of Restcomm. See "Appendix 10 - Collecting and Send Logs to TeleStax Support" at the pdf: http://telestax.wpengine.netdna-cdn.com/wp-content/uploads/2013/08/RestcommforVI-Advanced-User-Guide.pdf

Thanks

George

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[Bjørn Remseth]

Good.  The error I made was to have a <sip:sip:2234@... > address, and that screwed things up.     Now it works, and I can progress.   Thank you ;-)

kl. 22:20:20 UTC+1 mandag 28. oktober 2013 skrev Desi Pepova følgende:

[Gilberto Pereira dos Santos Júnior]

Hi guys, i have the same problem i guess. i do it , like that tutorial but :
i use jitsi with some samples number are there, it works very well all that ends with:
/restcomm/demos/dial/client/*.xml
I can call and listen easily.

but, all projects that ends with:
/restcomm-rvd/services/apps/*/controller,
include their own samples of app everything that ends with 'controller' never works.

someone can help me ?

[Desi Pepova]

Hello Gilberto!

In order to make a call to your application you
should dial: +your_app_number@IP_ADRESS:5080. It is not possible to
dial application URL. Also keep in mind that when you restart your
instance the IP address changes. It means the you should use the new
IP address while dialing, not the old one.

Regards!

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Desislava Pepova
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