Archives : March-2016
Title says it all – for the time being I am stuck deploying Asterisk in ESXi . We are also looking at Proxmox for our next round of servers.. Everything works fine except conference calls – very stuttery , have tried a few different codecs.I assume t..
!I want to install a Telephone IP Avaya 4610SW in my Asterisk 11, but Icant install.It asks a TFTP/HTTP Server, but is necessary I install it in mu Asterisk Server for works my Telephone?The manual is here https://downloads.avaya.com/css/P8/documents/003880182Tha..
My company has invested heavily in Counterpath’s Stretto provisioning platform for Mobile and Desktop VoIP clients .At this time their system allows 2 devices (for example iPhone + desktop computer) using the same software license per user , wh..
Im moving away from meetme to confbridge. the only remaining task i have is to convert (a ton of):exten => x,n,Page(SIP/123&SIP/124&…,diqA(mysound))or exten => x,n,Page(SIP/123&SIP/124&…,iq)(both inbound and outbound)I started down a very long r..
From what I can tell from the Wiki page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MP3Player, if someone dials in and starts playing a stream, mp3player will load up the URL and inject it into the current call.But what ab..
The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0:The ability to compile and run Asterisk with a bundled version of pjproject.Why would you want to do this?Several reasons: – Predictability:W..
I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient.I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicema..
Im building a CentOS 7 Asterisk and find my system log full of messages like this:Mar5 17:07:01 pbx2 systemd: Started Session 823 of user asterisk. Mar5 17:07:01 pbx2 systemd: Starting Session 823 of user asterisk. Mar5 17:07:11 pbx2 systemd: Remo..
I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip.In my snom 760 the setup for these two accounts is identical.When I call echo test from the account using chan_sip audio comes through fine.When I call echo test from the acco..
Hi!How can I setup my Chan Dongle recived calls in my Asterisk?I have to setup in dongle.conf ? Or in extensions.conf?And the code for recive I found this site http://asterisk-service.com/page/chan-dongle-useI have to To save Subscriber Number before?..