I have installed an asterisk it repeats this statement, again and again, my server is overloaded because of these errors.
[Sep 19 18:34:23] NOTICE[1641][C-00000398]: chan_sip.c:10995 process_sdp: No compatible codecs, not accepting this offer!
Some…
I have problems with setting up a webrts connection with sipml5 through an asterisk. When I check the status of https through the asterisk console, I get a response that it is up and running. But when I try to connect to sipml5 in the browser, I get…
I have planned to implement Asterisk SIP server for testing eMTA calls. I don't have eMTAs so I decided to start with Linux Soft client and then, when I will have an eMTA and physical access to the equipment play with eMTAs.
I installed Asterisk on…
I have an old PBX from Yeastar that contains an instance of Asterisk 1.6.2.6.
How can I enable the queue_log setting and have it persist across reboots?
I want to connect(for learning, nothing serious), two asterisk server via ISDN PRI.
The first server will act as "telco" so is pri_net, the second server is the cpe, so I will use pri_cpe.
The cards are
pbx1-net: Digium TE205P
pbx2-cpe: Openvox…
When building Asterisk, there is two different entities related to audio formats - formats and codecs. What's difference?
For example, I need a support only for alaw+ulaw+g.729 on voice traffic itself, and only plain wav + mp3 for announces and…
I have been trying to install Asterisk-18.10.1 version on my ubuntu(20.04.4) running inside VM. I was able to maintain connection from GoTrunk SIP endpoint and Zoiper as softphone. Followed https://github.com/GoTrunk/asterisk-config/tree/dynamic-ip…
I'm looking for a way to change pickupexten in Asterisk 11. Is there a way?
The features.conf:
pickupexten = *8
pickupgroup=1
pickupsound = beep
pickupfailsound = beeperr
I want to change *8 with any number. But when I changed it, It doesn't…
I'm trying to forward a specific incoming header to the other leg of the call, but can't figure out how to pass the value of the header in the incoming leg to the pre-dial handler
[addheaders]
exten => addheader,1,Verbose("Setting header")
exten =>…
am trying to run script in background which contain command such like "asterisk -rvd >> xyz.log", when I run it directly it work well but when run it with any background way (service, cron, &, nohub,,) it stop directly with this message:
stopped
so…
I have an Asterisk server (15.5, FreePBX) with three SIP trunks from different providers configured, two of them are working fine while the third for every call keep sendind the invite despite the correct answer from the trunk.
The trunk was working…
I just set up a server with Asterisk set up but when I try to make a VoIP call the line automatically drops...
I use CommPeak as a SIP provider and Bria for the interface
Here's my…
Incoming calls go straight through to failover regardless what other settings are. I have deleted the queue and rebuilt it, nothing seams to fix it...
Asterisk 16
Freepbx 15
Debian 10
There is a warning message "Unable to join queue" even though it…
I have set
load= chan_alsa.so
and i got this error
ERROR[77064] loader.c: Error loading module 'alsa.so': /usr/lib/asterisk/modules/alsa.so: cannot open shared object file: No such file or directory
is there a missing module or alsa is global ?
When Asterisk announces the position of a caller in the queue, it plays queue-thereare, followed by the caller's position, followed by queue-callswaiting. We have custom messages for each of these that work perfectly around the caller's position…