Archives : December-2017
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Ive been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this:Retransmitting #10 (NAT) to 185.107.94.10:13057:SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1—zg4cfkl50hpwpv4p;received5.107.94.10;rport057Fr..
I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this:[outbound]Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier))Ex..
Asterisk Project Security Advisory – AST-2017-014 ProductAsterisk SummaryCrash in PJSIP resource when missing a contactheaderNature of AdvisoryRemote CrashSusceptibilityRemote Unauthenticated Sessions Severity ModerateExploits KnownNo Reported OnDecem..
The Asterisk Development Team would like to announce security releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.These releases are available ..
Dear List It looks like the common way to to sip signaling over a trunk is: In the Request URI, return the Register Contact. In the To: Header, send the destination number. Unfortunately, asterisk with pjsip (i did not try chan_sip) does expect the dia..
The Asterisk Development Team would like to announce the release of Certified Asterisk 13.18-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asteriskThe release of Certified Asterisk 13.18-ce..
We just updated from 13.17.1 to 13.18.4 and are noticing a new error [2017-12-21 10:12:48] ERROR[32343]: res_pjsip.c:3850 endpt_send_request: Error 320055 DNS Refused (PJLIB_UTIL_EDNS_REFUSED) sending OPTIONS request to endpoint 6162480909.8009 The ..
We are proud to announce the first release 0.0.1 version of kannel-asterisk integration project. The goal of this project is to allow asterisk users to use kannel capabilities like SMS sending and receiving. Please visit https://asterisk-kannel.sourceforge…
Following the recent discussions about CentOS, I am interested to hear whether anyone is successfully running Sangomas Wanpipe (for an analog telephony card) using CentOS 7? Upgrades of Wanpipe have been tricky at best in the past, and I dont want..
I am looking to configure asterisk queues in off-hook mode, that is, the agent calls into the system and stays connected to this call, when new customer calls, he is redirected to the queue which should distribute to connected agents.is this possi..