I have seen the following scenario that may lead to a corrupted and possibly invalid configuration file after using UpdateConfig through AMI, at least with Asterisk 11.25:There is a configuration file a.conf that contains several sections and also conta..
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_UpdateConfigAccording to the link above, the NewCat action allows for options that specify whether the newly created category is a template, and whether it should inherit from an exist..
I have a working telephone project that uses SIP.js 0.7.5 with Asterisk on the server side. Currently it handles both audio and video correctly.The SIP.js webpage has instructions for setting up a datachannel through a SIP call. The online demo u..
I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed in commit 8653da4fa228e1e289e09e5d024e11d24da87d..
I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the ..
I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue.My setup is as follows:Server:CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146asterisk-11.2..
Is there a way to use AMI to detect whether an agent that appears to be free is in its wrap-up-time period?I am using AMI to query the queue status and its members, in order to generate calls directed to the queue, and I do not want to originate ca..
I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfi..
I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution.Background: this client is a telemarket..
I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support requi..