I have patched astcc.agi with the HUP patch, but it still hangs from time to time. Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running Linux on 2006-05-07 00:31:09 UTC bye Ronald
Nestor A. Diaz
2008-Apr-16 12:45 UTC
[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Hello Asterisk People, I have two annoying bugs in asterisk, that i want to know if some of you have already found a way to fix: Background: Asterisk 1.4.18.1 Debian package back ported to Debian etch. 1. I use a queue with just on sip device, one call at a time, however and without reason just after some couple of hours the sip device show in use and then no calls are transfered from the queue to the sip device, i do a sip show inuse and this is the result:asterisk -rx "sip show inuse" * User name In use Limit 200 0 3 * Peer name In use Limit 200 1/0 3 Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, recreate 200 extensions and reload sip.conf Not so nice thing to do.... 2. AgentCallBack I know i shouldn't have to use this function, since it is deprecated but lets comment the behavior Everything works fine, but when there are calls waiting in the queue, and the agent log in using this function, the agent is able to take the call , but the system log off immediately after the agent hang up the call. No solution at the moment, just login in and log in until there are no waiting calls, for the agent to not be kicked off. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia
Mojo with Horan & Company, LLC
2008-Apr-16 16:26 UTC
[asterisk-users] Two annoying bugs of asterisk ( sip in use and agentcallbacklogin)
Nestor A. Diaz wrote:> 1. I use a queue with just on sip device, one call at a time, however > and without reason just after some couple of hours the sip device show > in use and then no calls are transfered from the queue to the sip > device, i do a sip show inuse and this is the result:asterisk -rx "sip > show inuse" > * User name In use Limit > 200 0 3 > * Peer name In use Limit > 200 1/0 3 > > Simple workaround: delete sip 200 entry from sip.conf, reload sip.conf, > recreate 200 extensions and reload sip.conf >Does a simple sip reload work, or do you really need to go to all the trouble of removing the peer definition?