Archives : April-2018
list, Hope you are all doing fine!I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of resea..
Hey All,So one of the jobs that I get to do as head of the Asterisk project is to help inform people about the yearly conference we have about Asterisk named Astricon.For those who are not familiar with it, AstriCon is a fantastic event for anyone t..
I dont know if this list is the best place to ask such question but here it is, anyway.In page [1], I can read in PJSIPs endpoint section configuration reference:identify_by username,locationWay(s) for Endpoint to be identifiedThen clicking over identify..
2018-04-27 14:59 GMT+02:00 Joshua Colp :Adding a type=global line solved this issue as I missed the comment;type=; Must be of type global (default: ) two lines bellow.Thank you very much for corr..
HelloIve just discovered this [1] invaluable blog post (thank you very much Richard for writing it) and its reference to PJSIPs endpoint_identifier_order setting.On my Debian Stretch box powered with a packaged Asterisk 13.14.1, I edited a pjsip.c..
From [1], you can read:If you dont have an identify section defined, or else you have res_pjsip_endpoint_*identifier_ip* loading *after* res_pjsip_endpoint_*identifier_user*, then …To remove the above uncertainty coming from modules loading ord..
Im setting an Asterisk 13.14.1 box (Debian Stretch with packaged Asterisk)to implement SIP trunking services ie to both trunk with carrier trunks and IPBX trunks from various brands.For various reasons, I was inclined to implement this services w..
Hi. i am looking for a way to have headers for each section of the Master.csv eg call duration, hangup cause, destination,… is there a way to add it and be there permanently, even after log roratation due to size or dat..
A while back (last year maybe?), there was a Digium blog post on setting up WebRTC.I was never able to get that working.I was working with Asterisk 15 on a RHEL derived distro and had no idea of where to go to shoot the failure.Has anyone got a tutor..
Dear Asterisk Community, For the past 24 hours or so, Digium’s upstream provider has had a few outages that have affected Asterisk community services, including Asterisk.org, the mailing lists, and potentially other services.We apologize for any inconvenie..