Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10] ERROR[7664]: chan_sip.c:3030 update_call_counter: Call to peer 'rehmat' rejected due to usage limit of 2 -- Couldn't call rehmat == Everyone is busy/congested at this time (0:0/0/0) Im using asterisk1.4.0 . declaring type=peer solves the problem. but if anybody knows why its not working for type=friend, plz share. -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070317/c67aad03/attachment.htm
Tomislav Parcina
2007-Mar-18 12:02 UTC
[asterisk-users] Re: Call counter for sip misbehaving
Rizwan Hisham wrote:> Im using asterisk1.4.0 . declaring type=peer solves the problem. but if > anybody knows why its not working for type=friend, plz share.Please try 1.4.1, this should be fixed. -- Tomislav Parcina firstname.lastname@email.t-com.hr
17 mar 2007 kl. 12.21 skrev Rizwan Hisham:> Hi, > I have declared my sip users call-limit=2 and type=friend. When any > user recieves a waiting call while already in a conversation, the > peer call counter is set to 2.The problem is that, the counter is > not reset to zero after hangup and becoz of this the user is not > able to recieve any call anymore even if s/he has hungup. the > asterisk cli displays the following error. > > [Mar 17 16:15:10] ERROR[7664]: chan_sip.c:3030 update_call_counter: > Call to peer 'rehmat' rejected due to usage limit of 2 > -- Couldn't call rehmat > == Everyone is busy/congested at this time (0:0/0/0) > > Im using asterisk1.4.0 . declaring type=peer solves the problem. > but if anybody knows why its not working for type=friend, plz share. >Have you read sip.conf.sample? ;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time ; from the phone to asterisk ; 1 for the explicit peer, 1 for the explicit user, ; remember that a friend equals 1 peer and 1 user in ; memory ; This will affect your subscriptions as well. ; There is no combined call counter for a "friend" ; so there's currently no way in sip.conf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. We do document things now and then and expect users to read the documentation :-) Also please check http://lists.digium.com/pipermail/asterisk-dev/2007-February/026190.html /Olle