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Audio drop out at the beginning - Asterisk SIP - Asterisk ...
https://community.asterisk.org/t/audio-drop-out-at-the-beginning/75431
Hi, This post relates to my previous one on firewall rules (How to limit SIP registrations). To tighten security I set up iptables INPUT policy to DROP and allowed all expected traffic to go to my Asterisk server. Initially, I had the INPUT policy set to ACCEPT which was the default during the installation. Running Raspbx behind my ISP gateway and the gateway only …
After answering, the call is dropped - Asterisk SIP ...
https://community.asterisk.org/t/after-answering-the-call-is-dropped/84247
After answering, the call is dropped. Asterisk Asterisk SIP. onurcomert60 May 24, 2020, 4:52am #1. Hello there, My system is like this: Centos 7, asterisk 16.6.0, freepbx 14.0.10.3. The server is not behind nat on wm. It has an real ip. The call is answered by a queue agent, but sometimes it is suddenly dropped later.
Intermittent Audio Drop-Outs (and maybe dropped calls ...
https://community.freepbx.org/t/intermittent-audio-drop-outs-and-maybe-dropped-calls/9388
The phones are Cisco SPA504G for deskphones and Snom M3 for wireless. There are three Linksys ATAs (one on each PBX). As far as we can tell we have zero problems with faxing (surprise). For the most part it works well. However at least once per day there are a couple of audio fades/drop-outs and/or a dropped call or two or three.
Correcting One-way Audio with a VoIP Call - Asterisk
https://www.asteriskpbxsystems.com/troubleshoot-oneway-audio.html
Connect the ATA/IAD or other SIP device directly to the first device on the LAN such as the modem. Check for normal operation and good two-way audio. If you experience one-way audio or do not receive dial tone, then the issue will most likely be SIP unfriendly NAT, or a firewall present that is not allowing the correct VoIP packets to cross.
SIP with NAT or Firewalls - Asterisk Guru
http://asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
Asterisk will send the audio to the port and ip where its receiving the audio from. Instead of relying on the addresses in the SIP and SDP messages. This will only work if the phone behind nat send and receive audio on the same port and if they …
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