Archives : August-2022
I believe I have everything configured correctly, but Asterisk is complaining about my configurationIt is complaining about confidence settings.From the Asterisk Geolocation Implementation Wiki, I believe I have this set correctly.Sub-parameters:* val..
We have instances where we dial multiple destinations simultaneously and then an answer macro prompts the callee to press 1 to accept the call or 3 to reject. Previously if they pressed 3 (or just hung up) the other destinations would continue to ri..
We have an Asterisk 13.38.2 server which today started giving we couldnt allocate a port for RTP errors. The output of netstat -anp showed that Asterisk was using all 10,000 ports allocated for RTP, even though it only had a maximum of around 200 concurr..
Looking at the Asterisk wiki https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+ImplementationI see the dial plan support the GeolocProfileCreate and there is support for GEOLOC_PROFILE settings to be set on the dial plan.We currently ..
I realize that the heyday of DUNDi was about 2008, and that theres less and less information online about it and lots of people dont use it anymore and use static IAX trunks instead. But we have 53 asterisk phone systems connecting our locations, ..
We have an Asterisk dial which sends the call via a proxy using //, for example:Dial(SIP/${EXTEN}@peer_address//proxy_address)Does anyone know how we can make the SIP to the proxy use TCP? We tried making proxy_address match a peer in sip.conf with transp..
Hi. I am using Asterisk 16.27.0 in FreePBX 15.0.23.11. I installed via the FreePBX ISO (SNG7-PBX-64bit-2104.iso). I used the GUI to enable the Opus Codec in Asterisk SIP Settings and I can confirm the calls are using Opus 16000. How can I turn on f..
I want to reload the dialplan via the AMI, and I found the documentation showing the command: Action: ReloadActionID: Module:And the module names are* cdr* dnsmgr* extconfig* enum* acl* manager* http* logger* features* dsp* udptl* indications* cel* plc..
Subject: Asterisk IP PBX VoIP Servers Hacked by Hackers Good day from Singapore, I am sharing the following news articles for more awareness. News article #1: Hackers Targeting VoIP Servers By Exploiting Digium Phone Software Link: https://thehackernews.com/2022/07/hackers-targeting-voip-servers-by.h..
We have an Asterisk server with 3 IP addresses, and need to listen on only2 of those. This is with chan_sip. Does anyone know if its possible?If Asterisk listens on the third address then it seems to cause problems with the media address put in the ..