Archives : June-2016
Could someone kindly explain how does least recent strategy work?According to the config:leastrecent: rings the interface that least recently received a callThat does not explain much in detail. What happen if agent been idle (pause member) or in w..
im creating an outgoing call to number xxx with this command:http://host:port/mxml?action=Originate&Channel=Local/xxx@to-external&Exten=testDTMF&Context=cRETEUNICA&Priority=1wich points correctly to this portion of dialplan:[cRETEUNICA]exten => testDTMF,1,Ans..
im using an oldAsterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this:1) receive a call and put it on-hold in a queue (OK)2) monitor the queue and trigger an outbound call to a remote number using AMI, sett..
https://www.mundoopensource.com.br/yealink-t21p_e2-com-bugs-no-firmware-bugs-in-the-yealink-t21p_e2-firmware/Regards,Marcelo H. Terres IM: mhterres@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/mar..
I am in process developing Multi-Tenant system for Call Centers.I am considering what are the best option for Agent to Login and and wait for the calls from the Queue.Option 1: AgentLogin (staying on the line with music on hold and bridging the c..
I dont understand what a SIP invite is.Certainly its explained as:This article explains the main fields included in a SIP INVITE, which is sent to set-up a VoIP call. A SIP INVITE message contains typically between4 and 6 header entries with cont..
Im trying to use Asterisk 13.9.1 with Homer SIP Capture Server.My hep.conf Asterisk configuration is:[general]enabled = yes capture_address7.170.151.154:9060;capture_password = foo capture_id = 2464SIP Signaling work correctly but no RTCP STATS arr..
,My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP Users secret. But the voip engineer before me didnt save / documented those password. Now the servers hardware is begin to broke, it hangs a lot, and have a lot..
I am using Asterisk 13.9.1 and want to catch AttendedTransfer, but it is not fired at all.Thank ..
We use Asterisk extensively for conferencing – for the last 8 years or so this has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 350 concurrent users. Right around that number DAHDI hits a hard coded memory limit and ki..