I have installed and configure asterisk on my server, everything is working fine, but the problem is when user connected first time following message appears on sip debug :
[May 27 22:13:25] WARNING[20193]: chan_sip.c:3727 __sip_xmit: sip_xmit of 0x7f16b400bdb0 (len 646) to 192.168.0.150:61709 returned -1: No such file or directory
Scheduling destruction of SIP dialog '32729691-cce2-c278-d528-924721cfddca' in 32000 ms (Method: REGISTER)
but on refreshing page sip peer is reachable.
Seems like communication is one way only. The asterisk receives, but cannot send. Check firewalls, NATs, routing.
I have solved the problem, I have an error in my sip.conf file, changing sip.conf solves the problem
Related
I am using Perforce as part of a small development team. Everyone was able to connect to the P4V client except for one person who gets the following error:
TCP receive failed.
read: socket: WSAECONNRESET
We have deactivated his McAfee firewall and virus scan, but the error persists. I really don't know what to do with this error and it seems to be rather undocumented on the perforce website. From what I gather, it's because it's not a perforce-specific issue, but rather a TCP communication problem that might be caused by something else.
Any tips?
a TCP communication problem that might be caused by something else.
This is possible, or it's possible that whenever this user connects it causes some sort of server fault.
https://msdn.microsoft.com/en-us/library/ms740668.aspx
WSAECONNRESET 10054 Connection reset by peer.
An existing connection was forcibly closed by the remote host. This normally results if the peer application on the remote host is suddenly stopped, the host is rebooted, the host or remote network interface is disabled, or the remote host uses a hard close (see setsockopt for more information on the SO_LINGER option on the remote socket). This error may also result if a connection was broken due to keep-alive activity detecting a failure while one or more operations are in progress. Operations that were in progress fail with WSAENETRESET. Subsequent operations fail with WSAECONNRESET.
Beyond the usual connection troubleshooting questions (is this user on the same subnet? same version of the client software? same exact P4PORT setting? is the user able to connect via the command line client and if not does it give a more helpful error? why is this user unlike all other users?) I'd look at the server logs to see if it's logging any sort of more helpful error when this user tries to connect.
I am using Apache Jmeter to send FTP load on a Server. After setting up the FTP request on specific IP and port, I get this error:
Requested action aborted: Access violation at address 005F6DB2 in module '***.exe'. Read of address 00000000
I know that I have all the access and I don't know why a module can block me, although I know that software (***.exe) is not blocking my ports. What is the problem?
Here is the attached Wireshark screenshot from the server when sending the clients request, the red line is the problem and it occurs in different places each time I run the test.
The problem doesn't seem to be related to JMeter
Try uploading file(s) using "normal" FTP client like FileZilla or WinSCP.
If the problem persists:
try identifying its cause using Windows Event Viewer
try to trace system calls using i.e. WinDbg
or just raise an issue on your application (FTP component)
If the problem does not occur on "real" FTP clients:
double check that your FTP Request sampler configuration is correct, see Load Testing FTP and SFTP Servers Using JMeter guide for details.
try uploading the file to the other, i.e. public FTP server to see if it works
try implementing file upload purely in Groovy using JSR223 Sampler. See FTPClientExample.java for the code you could re-use. Make sure you have the following line in your script:
ftpClient.addProtocolCommandListener(new PrintCommandListener(new PrintWriter(System.out), true))
I've just setup FreePBX on my VPS. Everything is OK there are no notices. There hasn't been any errors in the installation. When I create my extension from the FreePBX create new SIP extension and try to connect afterwards I get Forbidden on my SIP client. The extension and the password are the same as I setup in the PBX.
This is the message from the log:
[2014-11-25 23:53:49] NOTICE[9209] chan_sip.c: Registration from '<sip:777#$$SERVER_IP$$>' failed for '$$MY_IP$$:59096' - Wrong password
The strange thing is that when I create an extension from the sip_custom.conf I connect to the server successfuly.
Any ideas :?
You have do apply changes(red button) after each change
You also have ensure your nat settings setuped correctly
You can check that device added in sip_additional.conf file and in asterisk via
asterisk -rx "sip show users"
Again, if you use nat=no and your device after nat, it will not work. Same can be if you use nat=yes and you have no nat.
I have installed asterisk I want to test I'm able to log in. I have in Go GoGrid. when I setup the IP, the extension and password I get it cannot reach the server. I'm able to ssh into the machine.
Is there a way to test from the console that asterisk is running and receiving connections ?
I would like to test from the terminal that I can connect with that extension and password ?
There might a problem with Asterisk running on GoGrid ?
Thanks,
Federico
Federico. Firstly, I run four Asterisk PBXs on GoGrid, so I know it works just fine.
Make sure your IP Tables is allowing SIP, IAX2 & RTP connections; by default, the images supplied don't.
To see if Asterisk is "seeing" a connection, use "asterisk -rvvv" to connect to the running Asterisk console. Use "sip show peers" or "iax2 show peers" to see if your phones and trunks are connected properly.
I also find that the "iftop" utility is -very- useful for making sure that remote devices are even trying to reach your PBX eth0.
from ssh you can go asterisk console by
asterisk -r
to check inbound sip packets come via firewall use
asterisk -r
sip set debug on
more info can be found at asterisk debug page on voip info
Also you can check whether Asterisk is alive by a console SIP client such as Linphonec and pjsua.
http://www.linphone.org/eng/documentation/guide/linphonecsh-control.html
http://www.pjsip.org/pjsua.htm
I am a newbie for Asterisk, so please be patient.
I would like to perform phone call originating by my application via Asterisk. I was recommended to use skype connection to terminate the call.
So, I installed AsteriskNow on VM with CentOs, created Skype Business account, got SIP ID and configured sip.conf and extensions.conf and explained here
http://forum.skype.com/index.php?showtopic=487451
Then I restarted asterisk service and checked log file. I did not see any error messages, so I hope that the configuration is accepted.
Now the question is: what do I have to do now? I want Asterisk to dial some phone number. It should arrive to skype that should forward the call to phone via VoIP gateway.
How can I do this?
Have a look at my answer: Asterisk click to call
You can do similar thing. At first do test with your local VoIP clients. You can probably install 2 SIP soft phones, configure them in sip.conf and using CallFile test if you can make connection between them. If this work, then instead of using Extension: SIP/test1 change it to valid dialable Skype for SIP "number".