Calls stuck in BSS with the status "Accepted"

Discuss any topic about Brekeke SIP Server.

Moderator: Brekeke Support Team

Post Reply
achooi
Posts: 21
Joined: Mon Aug 17, 2009 2:48 pm

Calls stuck in BSS with the status "Accepted"

Post by achooi »

1. Brekeke Product Name and version: BSS Version 2.3.8.4 Standard

2. Java version: 1.5.0 (build 1.5.0_07-b03)

3. OS type and the version: Windows Server 2003 Ent. Edition. SP2

4. UA (phone), gateway or other hardware/software involved: pap2

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Pattern 4

6. Your problem: I have many calls in my Active Sessions tab that shows the status "Accepted". My BSS is used to relay media and sip traffic to my asterisk servers. On the asterisk side, the calls complete and billing is correct. But for some reason, when the call is not properly removed as an active session on BSS. The reason this is a problem is because I need an accurate assessment of how many active sessions are on the BSS, with all the "Accepted" status calls, they are included in the overall active sessions. What could be causing this? Here is my dial plan:

Matching Patterns:
$request=^REGISTER
To=sip:(.+)@x.x.com

Deploy patterns:
$auth=false
$action=register
$target=x.x.x.x
james
Posts: 501
Joined: Mon Dec 10, 2007 12:56 pm

Post by james »

Do you mean there are remained sessions as "Accepted" even if these sessions are closed?

Is your SIP Server behind NAT??
achooi
Posts: 21
Joined: Mon Aug 17, 2009 2:48 pm

Post by achooi »

Yes, even if the call is closed on the Asterisk side, the call remains in BSS as "Accepted", sometimes as "closing". My sip server is not behind NAT.
james
Posts: 501
Joined: Mon Dec 10, 2007 12:56 pm

Post by james »

Hi
Do you have any other DialPlan rules?
For example, do you have DialPlan rule to disable RecordRoute by "&net.sip.addrecordroute=false" ?

Does this problem happen with a specific SIP client?
achooi
Posts: 21
Joined: Mon Aug 17, 2009 2:48 pm

Post by achooi »

No, I dont have any additional rules and most of our customers use PAP2.
james
Posts: 501
Joined: Mon Dec 10, 2007 12:56 pm

Post by james »

Hi

If you use another SIP client such as Xlite, does the same issue happen?

When did the issue start happening??
Post Reply