Archives : January-2023
Using Asterisk 18.12.0, a little confused on how to configure my pjsip.conf file to determine the codec to use for a call I have 2 endpoints:[Alice]disallow:all allow:ulaw,alaw,g729[Bob]disallow:all allow:ulaw,alaw,g729Alice calls into Asterisk on ..
We have used AMI for many years and Im in the process of migrating to ARI.My understanding is the call should be handed off to Stasis for the ARI application to control it.I was playing around with things and discovered the ARI hangup (DELETE /channels/{channelI..
Have you tried using the EVAL..
A couple years ago, I know I had ARI externalMedia working.Trying to figure out what Im doing wrong today.https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARIMy ari.conf[general]enabled = yes pretty = no allowed_origins = *[MyApp]typ..
This is just a test of the asterisk-users mai..
The Asterisk Development Team would like to announce the release of Certified Asterisk 18.9-cert4. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asteriskThe release of Certified Asterisk 18.9-ce..
there are new versions of Asterisk but mailing list is empty
http://lists.digium.com/pipermail/asterisk-users/
..
I am using a Swiss VoIP provider called sipcall. They have what they call a SIP trunk, and it is less expensive than individual accounts. From Asterisks point of view, this is just a regular SIP account, which can however receive and send calls f..
Hi. I have a very old dialplan (ie: a dialplan for a very old version of Asterisk) which Ive just transferred to Asterisk 16.28.0 The [globals] section of that dialplan includes: Kphones=SIP/KC470IP&SIP/KSnom870 Sphones=SIP/SYealinkT38G&SIP/SGC61..
how can I find very easy out the sender IP of wrong und unwanted tries to login as an SIP device to my Asterisk. The SIP port is allready not 5060, and for unwanted access i use #iptables INPUT DROP [Dec 26 14:40:11] NOTICE[8239][C-00000ff8]: chan_sip.c:26..