Archives : June-2022
Home » 2022 (paged 9)
Im trying to figure out how blind transfers are supposed to work with ARI. When two channels are bridged together through ARI, and one of them performs a blind SIP transfer, two things happen : – a Local channel is instanciated and goes through the dialp..
Greetings all,For those who are unaware Atlassian has published a security advisory[1]for Confluence. Theres no firm details yet or a fix, so as a result the wiki.asterisk.org site has been taken down for now. Im not sure how long it will be down si..
Im trying to get a sense for how many video calls with the Confbridge can be active when dropping the incoming video with the confbridge setup.So its really just the main persons video is showing out to all the endpoints. So its a one to many kind..
T3V0IG9mIGN1cmlvc2l0eSwgaXMgdGhlcmUgYW55IGRvY3VtZW50YXRpb24gb24gd2hhdCBpcyBw bGFubmVkIGZvciB0aGUgR2VvbG9jYXRpb24vRTkxMT8NCg0KSXMgdGhlIHBsYW4gZm9yIEFzdGVy aXNrIHRvIGV4cG9zZSB0aGUgU0lQIGJvZHkgYW5kIGxlYXZlIGl0IHRvIGRpYWwgcGxhbiwgQVJJLCBBTUkgdG8gcHJvY2VzcyB0aGUgZGF0YT8NCkZvciBleGFtcGxlLCBtaW1lIHBpZGYreG1sI..
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..