Category : Asterisk Users
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We have a customer where their switch sends pidf+xml presence information in the SIP INVITE message.Does Asterisk process this pidf+xml information?Does it store this in a channel variable that a dial plan could access?If not, does it store present t..
Just ran into another weird issue… In Swiss Telephone Interconnection, ptime=20 is a requirement. So on our SBC we enforce the presence of ptime=20 to avoid issues. I have an asterisk with chan_sip in the LAB which behaves weirdly… Inbound SDP au..
Hi. Im trying to use different logging verbosity levels to get dialplan output into different log files, and theres clearly something I havent understood about how Asterisk does this… I have the following in /etc/asterisk/logger.conf: [logfiles] logtest.verbos..
I have come over a codec negotiation issue. A (asterisk) is sending in INVITE containing * opus (type 107) * g722 * alaw (type 8) B answers with 183 containing SDP * alaw a=sendrecv B then answer the call with 200 and NO SDP I suppose that result..
The Asterisk Development Team would like to announce the release of Asterisk 19.4.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 19.4.1 resolves an issue reported..
The Asterisk Development Team would like to announce the release of Asterisk 18.12.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 18.12.1 resolves an issue repor..
The Asterisk Development Team would like to announce the release of Asterisk 16.26.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 16.26.1 resolves an issue repor..
We are running an Asterisk 13 server which is having a strange problem, where on calls which are received from the PSTN and then forwarded out to the PSTN again there is no audio for the first 10 seconds of the call. At the 10 second mark audio sta..
Team Im working on a scenario, where the registrar offers multiple instances that can handle registration: _sip._udp.reg.example.com has SRV record 0 0 5060 reg01.example.com _sip._udp.reg.example.com has SRV record 0 0 5060 reg02.example.com It lo..
I have been using Asterisk 18.11.2. Just tried Asterisk 18.12.0 and am running into a problem with the res_pjsip_transport_websocket.Using Ubuntu 20I use a bash shell script to compile Asterisk with settings. I didnt modify any settings from Aster..