This is the whole story we ever experienced after compiling and sipxecs-setup: When log into GUI, we found both sipxproxy and sipxbridge were in "shutdown" status. When looking at port 5060 status, we found 5060 were used by freeswitch. --------------------------------------------------------------- [root@sipx02 bin]# lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME freeswitc 2135 freeswitch 31u IPv6 13913 0t0 UDP localhost:sip freeswitc 2135 freeswitch 33u IPv4 13914 0t0 UDP sipx02.glms.net:sip freeswitc 2135 freeswitch 35u IPv4 13918 0t0 TCP sipx02.glms.net:sip (LISTEN) freeswitc 2135 freeswitch 36u IPv6 13917 0t0 TCP localhost:sip (LISTEN) --------------------------------------------------------------- After killing freeswitch pid seizing 5060, we found sipxproxy and sipxbridge in "running" and we can make call now. --------------------------------------------------------------- [root@sipx02 bin]# lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME sipregist 6410 lxsj 30u IPv4 146128 0t0 TCP sipx02.glms.net:39512->sipx02.glms.net:sip (ESTABLISHED) sipXproxy 11606 lxsj 6u IPv4 142950 0t0 TCP sipx02.glms.net:sip (LISTEN) sipXproxy 11606 lxsj 7u IPv4 142952 0t0 UDP sipx02.glms.net:sip sipXproxy 11606 lxsj 12u IPv4 142963 0t0 TCP sipx02.glms.net:sip->sipx02.glms.net:39512 (ESTABLISHED) sipXproxy 11606 lxsj 19u IPv4 146129 0t0 TCP sipx02.glms.net:sip->sipx02.glms.net:39513 (ESTABLISHED) sipXproxy 11606 lxsj 24u IPv4 146197 0t0 TCP sipx02.glms.net:39513->sipx02.glms.net:sip (ESTABLISHED) sipXproxy 11606 lxsj 33u IPv4 146418 0t0 TCP sipx02.glms.net:37201->10.10.1.5:sip (ESTABLISHED) --------------------------------------------------------------- It looks we lost freeswitch but in "Service" freeswitch still in "running". When we use softphone (x-lite) to call another softphone and turn into VM, we dial "0" for transfer to operator but no annoucement yet. The following is sipxvir.log capured : --------------------------------------------------------------- [root@sipx02 sipxpbx]# tail -f sipxivr.log "2012-10-08T09:36:08.905000Z":9:sipXivr:INFO:sipx02.glms.net:Thread-10:00000000:sipxivr:"SipXivr::run Accepting call-id NTBjODMzYmFlMzI5YjY5MzgyZWNjYmY2Mjg1NTk2Zjk. from [email protected] to [email protected]:15060" "2012-10-08T09:36:08.906000Z":10:sipXivr:INFO:sipx02.glms.net:Thread-10:00000000:sipxivr:"SipXivr::run Bridging the call" "2012-10-08T09:36:08.966000Z":11:sipXivr:INFO:sipx02.glms.net:Thread-11:00000000:sipxivr:"SipXivr::run Accepting call-id 6e61cecc-8bce-1230-0081-001a4beed718 from [email protected] to [email protected]:15060" "2012-10-08T09:36:10.766000Z":12:sipXivr:INFO:sipx02.glms.net:Thread-11:00000000:sipxivr:"Starting voicemail for mailbox \"2002\" action=\"deposit" "2012-10-08T09:36:10.802000Z":13:sipXivr:INFO:sipx02.glms.net:Thread-11:00000000:sipxivr:"Mailbox 2002 Standard Greeting" "2012-10-08T09:36:10.803000Z":14:sipXivr:INFO:sipx02.glms.net:Thread-11:00000000:sipxivr:"Mailbox org.sipfoundry.commons.userdb.User@162522b Deposit Voicemail from \"2001\" <sip:[email protected]>" "2012-10-08T09:36:10.805000Z":15:sipXivr:WARNING:sipx02.glms.net:Thread-11:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en" "2012-10-08T09:36:21.824000Z":16:sipXivr:INFO:sipx02.glms.net:Thread-11:00000000:sipxivr:"Collect::start 1 100/0/0 mask 1234567890ABCD#*i" "2012-10-08T09:36:21.942000Z":17:sipXivr:INFO:sipx02.glms.net:Thread-11:00000000:sipxivr:"depositVoicemail Collected digits=" "2012-10-08T09:36:27.305000Z":18:sipXivr:WARNING:sipx02.glms.net:Thread-11:00000000:ResourceBundleMessageSource:"ResourceBundle [EmailFormats] not found for MessageSource: Can't find bundle for base name EmailFormats, locale en" "2012-10-08T09:36:27.323000Z":19:sipXivr:ERR:sipx02.glms.net:Thread-11:00000000:sipxivr:"SipXivr::run" java.lang.NullPointerException at org.sipfoundry.commons.freeswitch.Transfer.createCommand(Transfer.java:30) at org.sipfoundry.commons.freeswitch.Transfer.<init>(Transfer.java:24) at org.sipfoundry.sipxivr.eslrequest.AbstractEslRequestController.transfer(AbstractEslRequestController.java:105) at org.sipfoundry.voicemail.AbstractVmAction.transfer(AbstractVmAction.java:84) at org.sipfoundry.voicemail.Deposit.runAction(Deposit.java:121) at org.sipfoundry.voicemail.VoiceMail.voicemail(VoiceMail.java:53) at org.sipfoundry.voicemail.VoiceMail.run(VoiceMail.java:42) at org.sipfoundry.sipxivr.SipxIvrApp.run(SipxIvrApp.java:31) at org.sipfoundry.sipxivr.SipXivr.runEslRequest(SipXivr.java:61) at org.sipfoundry.sipxivr.eslrequest.EslRequestScopeRunnable.run(EslRequestScopeRunnable.java:24) at java.lang.Thread.run(Thread.java:679)
Date: Mon, 8 Oct 2012 10:35:17 +0300 From: [email protected] To: [email protected] Subject: Re: [sipx-dev] Auto-Attendant Operator Announcement Not Work After Compiling from Source Code On Sun, Oct 7, 2012 at 7:59 PM, Tony Graziano <[email protected]> wrote: If you hear a greeting but the transfer fails it means the gateway cant handle REFER or has not been told to do so (assuming it is capable). Grand streams are not something I would ever use as a gateway, as they have been shown to be less configurable and don't strictly adhere to standards. If you don't hear a greeting when dealing internally from a soft phone it is more likely to be a soft phone configuration. Again, you are not providing ANY configuration detail. You could also check sipxivr.log file to see if any activity / call reaches AA application George _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev/
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