Archives : December-2015
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I continued to see this errors in the logs:[2015-12-02 10:05:57] NOTICE[19949]: chan_sip.c:23277 handle_request_invite: Failed to authenticate device 100;tagcdeaf7how do I guard against this kinds of attacks? Also, to get the IP address from where t..
Does the ENUM service e164.org died? All queries that I do lead toNXDOMAIN. I feel that the base has simply been empty for several months. Maybe some old addresses are still resolved, but the newer ones are not.The website still works, but it seems t..
I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually reach the PBX, but for some reason, they are not caught by any of my extensions context.He..
All;I am trying to setup SNMP on an Asterisk 11 system on CentOS 6. I have the net-snmp packages installed and I made sure that SNMP support is compiled into Asterisk and that the res_snmp.so module is loaded. res_snmp.conf is configured like so: [general]subag..
When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc …Is there a way to force URI calls through the PBX? I have fo..
I am trying to set up a default outbound endpoint for my Asterisk 13.6.0PBX, and per https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip, I do in pjsip.conf:[global]default_outbound_endpoint=SillyEndpoint…[SillyEndpoint]type=endpointetc.Howev..
HiI have a 3 level nested while-endwhile loop in a macro that when the execution reaches endwhile, it is jumping out to the While at the caller macro.It shouldnt since the are instructions after the endwhile.– Executing [s@macro-call-from-outside:..
Reminder: speakers deadline this Friday, 27 November at 23:59 UTCWe have already received several really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak.Many other dev-rooms a..
Im not sure what is going on there but I wanted to mention that Asterisk 1.8 is completely EOL, there will be no further fixes, even security fixes. For new installations you should use Asterisk 13 which is the most recent LTS.https://wiki.asterisk.org/wiki/display/AST/Asteris..
I have a puzzling situation, and would be grateful for any insight.I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g.[from-siptrunk]exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321)N..