Archives : July-2016
Home » 2016 (paged 35)
I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of sipregs. With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs.Is there something similar in pjsip? How ..
everybody.Well, I know that this is not the purpose of the list, but I started a crowdfunding project to allow me attend AstriCon 2016, as a speaker.My talk Using Asterisk and XMPP to provide greater tools to your customers and your users was appro..
Asterisk 13.8Is CALLERID(all) supposed to wok for pjsip? When I do this:exten => 1234,Set(CALLERID(all)=Jon Doe )same => n,Dial(PJSIP/phone123, 30)I expect the callerid to be as set, but is always seems to be phone123, the name of the endpo..
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..