Hello!! I have asterisk 1.6.2.10 and whenever there are more than 60 calls queued the following problem ocurrs. The agents hangup the calls but the do not receive new calls for some seconds or even minutes. If I seek throughout the full log I encounter that the Bye message coming from asterisk to the agents failes to arrive. If I make calls between extensions what I see is the following: Extension A calls extension B. Extension A hangups. Bye message is send from extension A to asterisk. Asterisk sends an Ok message to extension A. After some seconds or even minutes, extension B receives Bye message from asterisk. Any help will be appreciated!! Also I see messages like: Line 71344: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: Scheduling destruction of SIP dialog '065e1e5e612fcbf947d4f9044dd8ce1c at XXX.XXX.XXX.XXX'in 6464 ms (Method: INVITE) Line 71347: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: set_destination: Parsing <sip:2249 at XXX.XXX.XXX.XXX:XXX;rinstance=b877e4c23d44a205> for address/port to send to Line 71348: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: set_destination: set destination to XXX.XXX.XXX.XXX, port XXXX Line 71349: [Nov 5 10:58:55] VERBOSE[19043] chan_sip.c: Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:XXX: Any body can tell m what these Scheduling destruction of SIP dialog messages mean? Thanks!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121112/7342a629/attachment.htm>