Archives : May-2017
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Asterisk list!Ive been facing some scenarios in my dialplan where I see the hextension being executed for Surrogate channels. For me, it is kind of a mystery what these Surrogate channels are… Icouldnt find good information about them… the sou..
Im facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package).Please note: PJSIP is a free and open source multimedia communication libr..
I am using the Asterisk REST API in order to establish a call to an endpoint and to send over a remote file (HTTP). The issue is that I am experiencing an audio quality issue. I have tried encoding the file differently, but everytime Asterisk is cutt..
I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace…First of all, I would like to recommend this short cookbook that offers recipes for tackling dialplan fundamentals, making and controll..
Think janitorial or security kind of thing where an employee goes from location to location.Theyre supposed to clock in when they get to a site using a phone at that site to prove the..
I have made a success installation from source of Asterisk 14.4.0 on Debian Jessie (8.7). And I am running the Asterisk server, with several extensions and dialplans, all working well.However I am struggling to get app_jack to run.In menuselect I ..
Im using Asterisk 11 and have a problem with when making call transfer on remote Asterisk.This dial plan below works when I make a call directly to remote asterisk dialing FXO on remote asterisk.exten => 4,1,Dial(${FD_L1},25,trw)exten => 4,n,GotoIf($[${DIALSTATUS}=BUSY]?vmail:line2)ex..
i have strange problem with asterisk 13.15.0+pjsip bundled/CentOS 7/systemd start scriptwe are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days agotoday i have problems with stopping/crash..
all,Its slightly OT, but hopefully someone can help. Im struggling with getting Cisco IP Phone 7942G to fail over to our secondary Asterisk server in the event of a primary failure.We recently bought a bunch of new Cisco 7942G phones, which now c..
I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match.Anyone ..