I connected to a different carrier "B" today to test. This new carrier"B" does not have the current issues of calls not closing that i have with current carrier "A" . I have contacted my carrier "A" to check why their calls are not closing. What bothers me is that this started happening after i ...

SIP side close, Cell Phone close, ISP side closes but looking at "Active Calls on BSS i see the session active hence the Billing software continue to bill my client. Funny thing that happened today. I called my brother in UK from Canada and i hang off my phone and he hangs off his phone, about 5min ...

I removed "$session=com.sample.radius.proxy.RadiusAcct" and put in the below but the calls did not tear down

Deplore Pattern
$auth=true
To=sip:%1@sip.sipo.com
$b2bua=true
$continue=true

Infact i never had that problem 3 weeks ago when i was using ver 2.1.6.6 untill when i was working on enhancement of url replacement and i was advised to upgrade to Version 2.2.6.2. Now i have left the "url replacement" since this issue came on and sessions are not clearing/tearing down. I tried to ...

wayne106, Did you ever get a solution for this ? I am having similar issues where the session timeout is not working even with RTP-Relay "on" or "Auto". Some calls are not tearing down. I have done lots of stuffs even usning $b2bua=true and &net.sip.addrecordroute.lr , &net.sip.addrecordroute etc

Harold, You can open a new post for this. However if you set AUTHENTICATION -> INVITE in the Config page "ON it means that all devices must authenticate. However if you set "$auth=true" in the Dial Plan it explicitly means a registered device is only allowed to use that Deployed Pattern. see this ...

I am usning the "$auth=true" because of my Radius AAA Server. If i set "$auth=false" then unregistered callers will make free calls......lol

Thanks JanP,

I have it set as below and i APPLY the rule. however the issue remains

Deplore Pattern
$auth=true
To=sip:%1@sip.sipo.com
$session=com.sample.radius.proxy.RadiusAcct
$b2bua=true
$continue=false

I use "siptraffic"

I added them "&net.sip.addrecordroute=false and &net.sip.addrecordroute.lr=false" while troubleshooting. They have been removed and problem of calls not closing still exists. (PBX restarted after all changes) The current Dial PLAN $request=^INVITE To=sip:(.*)@ Deplore Pattern $auth=true To=sip:%1 ...

I have read the DialPlan tutorial guide over and over again and i dont know how to create the replacement dial plan for two different call legs "url-replacement". That is why i came to the forum for help.

Thanks James,

xxx always change. It could be any phone number

If i use "From = sip:xxx@%1 " it will only show what i define as "xxx" (Am i wrong ?)

1. Brekeke Product Name and version: Brekeke SIP Server for Brekeke PBX, Version 2.2.6.2 2. Java version: 1.6.0.100 3. OS type and the version: Win2003 Server Standard 4. UA (phone), gateway or other hardware/software involved: IVRCat, Cellphone 5. Select your network pattern from http://www.brekeke ...

I upgraded overnight and unfortunatly i could not see any [New Alias} page under [Dial Plan] I am using Standard commercial, i guess its only the Pro version that have that feature. Back to Square one. Let me re-define my issue again; Caller = xxx Callee = yyy IVR = IVRCAT LEG1 of the Call, xxx ...

I will upgrade at night and test tomorrow

I tested now and it did not work.

I tried to used the url you sent me. But i could not see any [New Alias} page under [Dial Plan]


Please go to the [Dial Plan] -> [New Alias} page.

I will have to test it tonight. However xxx and yyy always change and could be anything numbers. It is only 1111 and 2222 that always remains the same.

Yes i want to replace the From header "1111" or "2222" to show "xxx" when "yyy" recieved the call.

1. Brekeke Product Name and version: 2.1.6.6 2. Java version: 1.6.0.100 3. OS type and the version: Win2003 Server Standard 4. UA (phone), gateway or other hardware/software involved: RadiusCat and IVRCat 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html ...

I delibrately try to register a IP Phone that has no account

Thanks janP,

That solves the problem

1. Brekeke Product Name and version: 2.1.6.6 2. Java version: 1.6.0.100 3. OS type and the version: Win2003 Server Standard 4. UA (phone), gateway or other hardware/software involved: Grandsteam Phone, Linksys ATA 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...

That will only give me the log of the date i click. I want to be able to select 1st Nov 2008 thru 30th Nov 2008 and see all call logs within that period (which i can filter by caller).

1. Brekeke Product Name and version: 2.1.6.6 2. Java version: 1.6.0.100 3. OS type and the version: Win2003 Server Standard 4. UA (phone), gateway or other hardware/software involved: Grandsteam Phone, Linksys ATA 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...

1. Brekeke Product Name and version: 2.1.6.6 2. Java version: 1.6.0.100 3. OS type and the version: Win2003 Server Standard 4. UA (phone), gateway or other hardware/software involved: Grandsteam Phone, Linksys ATA 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network ...