Archives : May-2015
Home » 2015 (paged 44)
HiI want to load balance SIP calls between two(or more) Asterisks with only DNS SRV. I used bidirectional sync Unison to synchronize configuration files and internal database file between two Asterisk boxes.The problem is when a calls come to Asteri..
Hello.We have an issue with canseling dialogs.Scenario that we have issue is:Calling to some extensions from endpointHanging Up caller party until ringing send to asterisk from second leg(called party)Asterisk resend Bye to called party but Bye not go..
I need to leave an audio running in the background while other commands are executed. I have an AGI that does some checks and tamed around a5s, need to leave a audio running in the meantime. Does anyone know any way? already I tried hard and so far nothin..
Congratulations!Today is your lucky fucking day. Because youre about to discover a weird trick you use TONIGHT to get a woman so turned on and wet between the legs that SHE makes the first move.This free video shows you the secretYou can use this tr..
I noticed that a call on hold is disconnected after 5 minutes, whatever the value of the rtpholdtimeout parameter in sip.conf. Tested from v1.8.10.0 to 1.8.32.3. The version 1.8.8.0 is not affected. I dont know between 1.8.8.0 and 1.8.10.0.Does anyb..
Can anyone tell me how can I create echo test using ARI stasis ap..
HelloI have already several Asterisk servers running with similar configuration, but now I stumble into a problem.I have mysql configuration res_config_mysql.conf :[MyAsteriskDB]dbhost = 127.0.0.1dbname = MyAsteriskDBdbuser = astadmin dbpass = mysec..
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?or is chan_pjsip better supported?or the recommended way for asterisk is use respoke.io?my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)chan_sip.c:10496 process_s..
guysI have a site on Asterisk 1.8.11.0 running in CentOS 6.5 that has about 150concurrent callers.I keep getting these types of messages in the CLI:[May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap:Channel allocation failed: C..
, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, – — Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10..