Category : Asterisk Users
Home » Archive by category : Asterisk Users (paged 39)
contrib/scripts/get_mp3_source.shsvn export has to be changed to https else it fa..
Is it possible to change the callerid for specific agents logged in to a queue (local channels)? We can change the callerid before it hits the queue but we are interested in changing/blocking the callerid for specific agents. I dont see a way in ..
all,Ive got an old server (Asterisk 13.28.0) that Im trying to configure to store voicemail in a mysql database.I have sip realtime working via odbc and its been working well for years.However, when I recompile Asterisk in order to store voicemail..
Hellocan anyone explain to me why (and HOW) there is a difference in data between the Asterisk console sip show peers and the realtime MySQL configuration ?Using : asterisk-certified-13.21-cert6Asterisk console data :/usr/sbin/asterisk -rx sip show pe..
I know its been quite some time that the Asterisk community services have been down but Im pleased to say that most services are coming back online. DNS entries have been updated but they may take time to fall out of the various caches across the kn..
The Asterisk Community Services infrastructure (issues.asterisk.org, wiki.asterisk.org, gerrit.asterisk.org, crowd.asterisk.org, git.asterisk.org, downloads.asterisk.org, downloads.digium.com, signup.asterisk.org) is undergoing a physical move to..
The Asterisk Development Team would like to announce the release of Asterisk 18.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.3.0 resolves several issues repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.17.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.17.0 resolves several issues repor..
Hi. Im running Asterisk 13 and Asterisk 16 using SIP trunks only (to a commercial trunking provider).I have no analogue interfaces. A user reported dialling in to voicemail (the standard Asterisk Comedian Mail service) from a mobile phone and being una..
A call originated from ARI (using ari-py), changes state to UP while the called party is still ringing. The bearer is a PjSIP trunk. I am wondering if this is caused by any kind of early media or incompatibility between my Asterisk and remote SBC ..