Archives : February-2016
Greetings.I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile apps operating in environments that I do not control.I would like Asterisk to default to sending INVITES and all other SIP signals to endpoints via the exist..
everyone, I have some problems to enable push the Zoiper Windows Phone in my Asterisk 11.Below is the result of CLI == Using SIP RTP CoS mark 5– Executing [1033@ramais:1] Answer(SIP/1030-00000201, ) in new stack > 0x7efc90024190 — Probation pas..
I do it via a group count:main call handling:exten => sub123,n,Set(GROUP()122345)… the main routine calls subroutine:exten => general,1,GotoIf($[${busyonbusy}=YES]?100:200)exten => general,100,GotoIf($[ ${GROUP_COUNT()} > 1 ]?110:200)exten => general,110,Hangup(..
I found the app_swift module (that Ive been helping maintain) makes asterisk crash on versions higher than 11.19.0 – something that happened on 11.20.0-rc1 makes asterisk segfault.I realize app_swift is not a supported module — Im just having a h..
I have a phone in my living room (ext. 111), a phone in the kitchen(ext. 222) and a phone in my bedroom (ext. 333).Both phones are part of a ring group.exten => 7654321,1,Dial(SIP/111&SIP/222&SIP/333)Everything work fine and, as expected, all pho..
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..