Sockets Question

Hi, I’m working on a Java application implemented with AsteriskJava. I have a little flaw on executing OriginateCall. There are times that my mobile phone rings and there are times that isn’t. When it is, I can see on the logs that my Playback applications on the Dial Plan was executing already. I just wonder why this happens. On the Java logs,

Jan 29, 2009 5:36:58 AM org.asteriskjava.manager.internal.ManagerConnectionImpl connect INFO: Connecting to 192.168.236.128:5038 Jan 29, 2009 5:36:58 AM org.asteriskjava.manager.internal.ManagerConnectionImpl setProtocolIdentifier INFO: Connected via Asterisk Call Manager/1.0 Jan 29, 2009 5:36:58 AM org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin INFO: Successfully logged in Jan 29, 2009 5:36:58 AM org.asteriskjava.manager.internal.ManagerConnectionImpl doLogin INFO: Determined Asterisk version: Asterisk 1.4 Jan 29, 2009 5:37:00 AM org.asteriskjava.manager.internal.ManagerConnectionImpl disconnect INFO: Closing socket.

Is your phone registred with asterisk?
Have you looked up with tcpdump (for example)? Do you see packets with destination = your sip (java or what ever) client?
If you do, show asterisk log at the call time

Hi thanks for the reply. I’m not sure what are those tcpdumps. Here’s my logs on the CLI.

== Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 202.164.178.102 > Channel SIP/pynkglobal-sip-0079db10 was answered. -- Executing [6000@default:1] Goto("SIP/pynkglobal-sip-0079db10", "voicemenu-custom-2|s|1") in new stack -- Goto (voicemenu-custom-2,s,1) -- Executing [s@voicemenu-custom-2:1] AGI("SIP/pynkglobal-sip-0079db10", "agi://202.164.178.99:4573/mobiagi.agi?transactionid=202") in new stack == Manager 'admin' logged off from 202.164.178.102 -- AGI Script Executing Application: (Playback) Options: (mobiclear-welcome) -- <SIP/pynkglobal-sip-0079db10> Playing 'mobiclear-welcome' (language 'en') -- <SIP/pynkglobal-sip-0079db10> Playing 'beep' (language 'en') -- AGI Script Executing Application: (Playback) Options: (mobiclear-goodbye) -- <SIP/pynkglobal-sip-0079db10> Playing 'mobiclear-goodbye' (language 'en') == Spawn extension (voicemenu-custom-2, s, 1) exited non-zero on 'SIP/pynkglobal-sip-0079db10'
I’m using X-Lite to test the voice menu but running the Java aplication, I don’t use phone anymore. Is that necessary? Thanks

I am guessing by your Java logs that the asterisk-java server is closing the socket connection and NOT asterisk.

I would do a review of the java code to figure out why the manager connection is terminating.

Hi, thanks for the reply. Yesterday we moved on a new location. Data center to be specific with all the bandwidth we need. From there I notice that all calls are perfectly getting out. So I think the solution to my problem is getting a better internet connection. Thanks for your time reviewing my thread. :smile: