Archives : February-2016
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Ive ported an Asterisk 10 installation to Asterisk 13, and Ive noticed that whenever Asterisk plays my audio files it uses the slin format.I have not converted ANY of my audio files, which means asterisk must be converting my wav files to slin on ..
The Asterisk Development Team has announced the releases of:DAHDI-Linux-v2.11.1-rc1DAHDI-Tools-v2.11.1-rc1dahdi-linux-complete-2.11.1-rc1+2.11.1-rc1This release is available for immediate download at:http://downloads.asterisk.org/pub/telephony/dahdi-li..
Id like to transfer all my pesky telemarketing calls to Jolly Roger .http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.htmlIn the middle of a call Id hit some DTMF sequence, which would dial Jolly Roger and trans..
HiIm using asterisk 1.8.32.3 on CentOS 6Ive noticed when using queues that the members of the queue stop ringing for the duration of any set periodic announce. Is this the only behaviour possible or is there a way to set the members to continue ring..
I have an ARI application that is registered for Stasis in the dialplan. One of the events I reap in my application is a ChannelDtmfReceived. The thing is, Asterisk 13.6.0 sends me two DTMF for each DTMF pressed (have tried both SIP phones and landline..
I am having a problem trying to compile dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server.Version 2.10.2 compiles fine.Is there a new dependency for 2.11.0 that was not required for previous versions?Here are some of the errors I get: INST..
All,Im looking for a PSTN Card that I can use with my Asterisk Server to achieve the following goal :1. Detect FAX signal and route it to a specific extension.2. Detect an incoming call from the same PSTN line and route it to IVR.Do openvox FXO/FXS ca..
i have a issueAsterisk crash (Module res_odbc exactly) after the same log who is /ERROR[23805] astobj2.c: bad magic number…/you will see on the log :Today[2016-02-24 16:00:38] ERROR[23805] *astobj2.c: bad magic number 0x552f302e for 0x7fe3505b3958*[2016-02..
everyone! Everyday some channels go to this situation:# asterisk -rx pri show channels| head -n 32 | grep YesNo IdleYesPRI BChan Call PRIChannel Span Chan Chan Idle LevelCall Name11 YesNo Idle Yes12 YesNo Idle Yes16 YesNo Idle Yes1 13 YesNo Idle Y..
, I am trying to enable SIP SIMPLE communication in my test environment(Asterisk 13.6.0)I have two problems:1. Using messagesend(), I dont want my users to be able to change their own callerid name. I want the name that appears in the ${MESSAGE(fro..