Archives : April-2016
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I am trying to set-up an Asterisk server to transcode between different RTP frame sizes. I have devices on one side using alaw with ptime 20 ms and some equipment on the other side requiring ptime 10. I am using the latest Asterisk 13.8.0. The set..
all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc?I will be glad if it is an open source co..
What is the best virtual server tech (and most stable, etc) to use for a asterisk virtual hosting environment?I have a client that wants to do virtual hosting of Asterisk (only SIP or IAX, no PRI, etc) and Im wondering if Xen or something else wo..
I am currently having a voice quality problem with one of our Asterisk servers.We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions.I am looking at the timing source for Aster..
For lab testing, Im trying to build two differents PJSIP trunks between twoAsterisk 13.8.0enabled boxes. I thought I could set up both trunks like this:Box A/port 5060 Box B/port 5060Box A/port 5062 Box B/port 5062and declare trunks like this:[foobar1]type=endpo..
In a fork of seanbrights opus patch for 13 there are further patches for Forward Error Correction and Package Loss Concealment, both of which ought to very useful in voip:https://github.com/traud/asterisk-opusAnybody used these patches ? Puzzled ..
I get the following error when trying to update date the database via contrib/ast-db-manage/alembic -c config.ini upgrade head. Every previous update has always worked any idea what i..
everyone!Hurray, I found the problem!After reading http://stackoverflow.com/questions/11812731/first-udp-message-to-a-specific-remote-ip-gets-lost I made some more traces, and the problem is that my box loses the arp entryof its router. In a trace..
I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bri..
As you know, the ability to use a bundled version of pjproject was introduced with Asterisk 13.8.0.More info on the Asterisk Wikiand in this email thread.Since then Ive fixed a few issues related to older versions of Debian and CentOS which you can..