Archives : March-2019
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Hello! My name is Emma.I came to your city for a week. I stayed at the hotel and now Im bored. Maybe you come to me?On this site you can find my address.http://emmanicella.suAnd… see some of my p..
From my provider I get extensions of: +1 1 seemingly randomly. What Id like to do is exten=_!1234567890,1,Answer() which would match anything ending in 1234567890. But that doesnt work since ! can only be used at the end of a pattern. I tried [+\ ]..
Hello IVR call is coming. I want 8 (digit) in the loop. How can I do that ? Current confi: [ivr1] exten=>_XXXXXXXXXXXX,1,answer() exten=>_XXXXXXXXXXXX,n,background(/var/lib/asterisk/ivr/ob) exten=>_XXXXXXXXXXXX,n,WaitExten(10) exten=>_XXXXXXXXXXXX,n,Dial(${OPERATO..
What is CooMeet?Welcome to CooMeet – the next generation of video dating, for the user who wants to talk online to p..
Running a test using asterisk 16.1.1 and two PCs with Firefox browsers.Im running the cmp2k demo.I place calls into the same asterisk and using AMI answer the calls and then add them into the same confbridge. Video mode is configured to follow_talker.Howev..
Using asterisk 16.1.1.Im setting up a test using the cmp2k (Cyber Mega Phone 2K Ultimate Dynamic Edition).I have noticed Chrome 72 had some issues with video streams.I just upgraded to Chrome 73 and see they still have some issues.If I have 2 calls..
since ipv6 doesnt really push ipv4 out of our networks, we still use IAX (when possible) to not face NAT problems from SIP, Asterisk 16 included. Question is, what is the future of IAX. Is it a dead protocol ? Will it stay as is even only security probl..
Joshua Does the survey imply that there are big changes coming for Asterisk? E. g. features or facilities will be dropped / deprecated from the open source version innew releases, big changes to existing facilities / protocols, what is supported offici..
Lets say I have to make 40 phone calls quickly.If I use the AMI interface to originate a call, close the connection, open another connection etc… This works. but is slow…If I open the AMI interface and originate a call – DONT close the interfa..
I rey to register an Asterisk 16.2.1 pjsip to an ASTERISK 13.25.0 chan_sip using ipv6 and pjsip_wizard. I only got it work if in remote_hosts I put the ipv6 address and not the hostname like sip.domain.ltd No need to say that an AAAA entry is exist..