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..
Author : Benoit Panizzon
Gang We migrated our voicemail system from asterisk 13 to 16 a couple of months ago. Right after the migration, we got the complaint that vm-intro is being played when the customer had recorded a own announcement. So I assumed we had replaced that f..
We want to be able to reject some pjsip calls with a temporary failure so that the PBX of the caller plays an announcement in the language of the caller, that the call can temporarily not reach the destination. https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappi..
Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answer..
Gang I have not yet managed to find a solution to correctly generate CDRs for this situation: Alice calls Bob. Bob has call forwarding delayed 20s to Charlie. Charlie picks up immediately. exten => bob,1,DBget(cfwdly=CFDLY/${exten}); $cfwdly conta..
Gang To get our customers more information on how they registered I am looking for a elegant way to get an information like the CLI command: pjsip show endpoint [endpoint] I had a got on ARI, but that basically only returns the information if an endpo..
Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unl..
Gang Mitel PBX use options without username to monitor the connection. Therefore Asterisk PJSIP cannot match an unsername against an endpoint and prints a notice on the console. Is there a way to silence this kind of notice? I wonder if identify_by hea..
Gang I gave up on running asterisk with two interfaces without it mixing up the ip addresses. So I have removed one transport definition from pjsip.conf Now * keeps complaining: res_pjsip.c:3461 ast_sip_set_tpselector_from_transport_name: Unable to retri..
I have been pondering over a problem to use an asterisk server behind an SBC unable to successfully handle registrations. Now I observed something strange which I suspect might be a bug on the asterisk side. The SBC originates is register from Port 6..