maslym

These two fields are blank. Actually I did't change any settings on the agent group after I created it. I just used all default settings. Thank you.

Hi, We found another problem on pbxnsip. Whenever there are calls in the agent group, it will cause the CPU load used by pbxnsip to jump to 99%. I tested it on our test server with 2.1G processor running SuSE Linux. With versions 2.1.1.2211 and 2.1.2.2292, when there is a call coming into the agent group and waiting to be picked up, the CPU load used by the pbxnsip jumps to 99%. There is no such problem found on verion 2.1.0.2115. Any idea? Thank you!

The reason why I posted this topic is because the pbxnsip license we purchased is controlling the number of concurrent calls on the pbxnsip. If the system is checking the concurrent calls based on the number shown on Status->General page, that means we can use only half of the concurrent calls capacity we paid for. Can you please advise how the system checks the concurrent calls for the purpose of the license? Thank you.

Hi, I notice today that the number of concurrent calls shown on Status->General page is always 2x the number of actual calls shown on Status->Calls. This happens on both internal calls between extensions and external calls between extensions and outside PSTN numbers. Any idea? Thank you.

Is it possible to put it back in the admin mode in the future release? This feature is very useful for us as we have the pbxnsip license controlling number of concurrent calls and we need to keep an eye on it. Thank you.

Is version 2.2.0.2415 for SuSE 10 available?

Unfortunately it seems the new version for SuSE 10 is still unavailable. Anyway, thank you.

According to the above link, it says that in version 2.1 and higher, if the call is redirected from another incoming call, it will use the original number. I tried to use "$f $a <account>" and "$f $a " as Trunk DID and still it didn't show the original caller's phone number as caller ID for redirected call. I was wondering if this is a bug of version 2.1.0.2115.

Okay the problem is fixed if I change "Default PnP Dialplan Scheme" to something other than "User must press enter". But it seems that I have to change such setting for all domains on the same pbxnsip server because the call from the cell phone will still hit DISA menu on any domain with "Default PnP Dialplan Scheme" set to "User must press enter". For example, if I register the cell phone under any extension in domain1 and set "Default PnP Dialplan Scheme" to "North America (3-digit extensions [2-7]xx)". The call from my cell phone into domain1 will hit the auto attendant annoucement. But the call from my cell phone to any other domain will still hit DISA menu if "Default PnP Dialplan Scheme" of that domain is set to "User must press enter". So it seems that the pbxnsip will search all domains for the registered cell phone number, not just the domain that the call from the cell phone is coming into.

There is no such problem if I change back to use version 2.0.3.1715 (Unix) with the same set of configuration files. Also we are using Snom 4S proxy server to provide VoIP service and there is no such problem as well. So I don't think there is something to do with RFC3325. In fact I could see the caller ID of call coming into the pbxnsip on the call history page. So it is just that the pbxnsip replaces the caller ID with the caller ID of the trunk when it forwards the call out. Can you please tell me what's the difference between version 2.0.3.1715 (Unix) and 2.1.0.2115 (Unix) on handling the caller ID for redirected call? Thank you.

When I set up call forward on my extension to any PSTN number, the caller ID of the redirected call is the caller ID of the trunk instead of the phone number of the original caller. I didn't set up anything related to the change of the caller ID such as the parameter of the extension, tel:-alias or the Trunk DID. This problem happens on 2.1.0.2115 (Linux). No such problem found on 2.0.3.1715 (Unix). Any idea?

When I access Status -> Calls to check concurrent calls, the web page stops refreshing by itself. I tried this with different browser programs like Internet Explore and Firefox and got the same result. This happens on 2.1.0.2115 (Linux). No such problem found on 2.0.3.1715 (Unix). Any idea?

If I register my cell phone number under my extension, when I make a call from my cell phone into my own pbx on the pbxnsip server, I will hit the auto attendant annoucement instead of the DISA menu. But when I make a call from my cell phone into other pbx on the same pbxnsip server, I will hit the DISA menu instead of the auto attendant annoucement of that pbx. This really confuses me. This happens on both 2.1.0.2115 (Linux) and 2.0.3.1715 (Unix). Any idea?