Archives : December-2019
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The Asterisk Development Team would like to announce the release of Asterisk 13.30.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.30.0 resolves several issues repor..
For years Ive been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Aster..
Disclaimer: I know I should not be using chan_sipThat being said I am trying to force Asterisk to use tcp by doing Dial(SIP/1234::::tcp@1.1.1.1//2.2.2.2)I want that Asterisk should send out the packet using TCP via the SIP proxy2.2.2.2. When I do a de..
all,I want to send a text message to a polycom phone. I know how to create a call file – but that will call the phone and nothing happens till the phone is answered.How do I create a call file that will send a text message over SIP to the polycom phone..
I have a customer who wants me to send a DTMF on the calling channel if the called channel says any word. So I am using[my_gosub_routine]exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2}) same => n,Playback(hello) same => n,Return()[default]exten => _X.,1,NoO..
–_000_AM5P192MB01476934158D6709857E5FAF89500AM5P192MB0147EURP_Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: quoted-printableI am using an ARI dialer for my applications and since my last upgrade to Ver. 13.29.2 from 13.23…
Im moving asterisk to a laptop, so cant use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ?
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i have following topology PSTN – Asterisk —- internet —– router – jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router – public IP/private IP (NAT) jssip client – private IP – ..
This has a bit of a long explanation… ultimately the question is why adding a section to sip.conf made a difference to One Touch Record.Were implementing a recording toggle using the Record button on a SIPtelephone and Asterisks One Touch Record feat..
All.Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller?I ask because weve had increase in spam calls and Id like to simply play dead air or something rea..