I am having one of those days. We just replaced an old Asterisk 1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost everything is working except for some incoming calls directed to a Cisco SPA-8000. The PSTN trunk is SIP..
Author : Carlos Chavez
Home » Articles posted by Carlos Chavez (paged 3)
I thought I had found and answer to this question by using CALLERID(ani) but it seems that only works on versions prior to 12. On Asterisk 13 setting CALLERID(num) before dialing to an external trunk always changes CDR(src) to the number ..
Usually phone companies set the outgoing CallerID for you but recently we got control over that and are now setting the outgoing Calleir ID ourselves. My problem now is that the CDR will put the outgoing CID in the CDR instead of the extens..
I have a very strange problem with my queues today. When the agent answers a call they get the periodic_announce sound played to them. I have a periodic_announce set to 60 seconds and the caller does hear it if their call is not answered..
We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered..
I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation. No matter what I try I alw..
I am trying to get the Mega Phone demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode.I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu ..
I just tried to compile the latest Asterisk 13.8.0 and it stopped with several errors on pjsip.So FYI if you run the install_prereq script and then use ./configure –with-pjproject-bundled you will have the same problem because the prereq script insta..
Has anyone used Telynx as a SIP trunk provider? It works with chan_sip but it I seem to be having problems trying to set up a PJSIP trunk. I always get a 401 Unauthorized when they send me a call. I know my username and password are correct si..
I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: AGI Tx >> 200 result=99981 (timeo..