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Messages posted by: armachea
Forum Index » Profile for armachea » Messages posted by armachea
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Hi guys,

Here I am again for another episodes of my daily problem.

I have a sip application in a live environment, that is sending large number of invite request. the strange thing is all clientTransaction activities in my wildfly10.0 are not closing.

this is some activities in slee-management console |'images/activity.context.gif'%20/%3E%20Activities |

AC (-1c..36:-7278) 60 org.mobicents.slee.resource.sip11.wrappers.ClientTransactionWrapper
AC (186..a40:30fd) -144 org.mobicents.slee.resource.sip11.wrappers.ClientTransactionWrapper
AC (-1c..af0f:153) -1289 org.mobicents.slee.resource.sip11.wrappers.ClientTransactionWrapper

i have made sure to end all the activities related to sip with the following
if (aci.getActivity() instanceof ClientTransaction) {
log.warn("SipSbb abortAll terminating client transation 2");
ClientTransaction clientTransation = (ClientTransaction) aci.getActivity();
} else if (aci.getActivity() instanceof ServerTransaction) {
if (aci.isEnding()) {
log.warn("server transation is ending, skip terminating");
} else {
ServerTransaction serverTransation = (ServerTransaction) aci.getActivity();

but am still have unclosed activities in my slee-management-console.

Please guys how can i fixed this nightmares.
Okey guys I fixed the issues, from my side . it could have been fixed from the operator side as well, but in any case i did it in my jainslee code.

Basically for international call >>> had to had some digit in both number in order to make the call.

Sudan to Lebanon: (part A +0249113400007) to (part B +0096179197073)

And voila, this trick made the magic.
Hi guys I trust this message finds you well and safe.

I have created an application with to make call with sip, [(A-part) making a call to (B-part)] this is basically my application. Well, as they say good things never last..... Locally, the application is working without any problems and i received the green light to deployed the application in a live environment.

Locally i was just using a | wildfly10.1 container to host my SBB app | | restcomm8.0 sip server for sip communication | | xlite3.0 to perform calls|

But Now in the live environment, things have get out of hands | i have the network operator | | i have DIALOGIC wich convert SIP traffic to GSM traffic|

Guys my main issue is when making a call to same country i have no problem:

Sudan to Sudan: (part A +249113400007) to (part B +249113400008) :::: partB is receiving the call and viewing partA number as (+249113400007) on his device.

Sudan to Lebanon: (part A +249113400007) to (part B +96179197073) :::: partB is recieving the call in a different format which is totally wrong, and viewing partA number as (+113400007) without the country code (249) in the number.

I want to know this problem can be fix from my application or from the network operator side with (Translation and Transformation Patterns) ?



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