All the other configuration options are not external to Asterisk. This is where I went wrong. I tried tweaking and changing asterisk settings for months to no avail. If you have No Audio - It's to do with your ROUTER or FIREWALL. If you've configured the ports as I've shown above, there are only a few things left to check. 1. Internal firewall.
the W52P Dect Phone got with outgoing calls no audio. The Called Person hear everything, but the caller (W52P) hear nothing. The other direction works fine. Attached the pcap File from an call to an softphone (PhonerLite). The VoIP Server is an local Asterisk 13.1 I hope, that anybody can help me Thanks However, Asterisk does not understand ADPCM WAV files. To convert your WAV files to a format which Asterisk can understand, use the following command: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql. Note that sox v14.3.0 and above (installed in Ubuntu 9.10), resample is no longer used, remix is used or leave it as: I see many instances of no sound issues discussed around the place, but most of them concern nat issues which I don't believe is my problem. Setup is: Pots line – Asterisk (IP04 appliance) – X-Lite softphone. The Asterisk box and PC running X-Lite are on the same subnet & switch. Jun 15, 2015 · One way audio is a common issue that we’re often called upon to troubleshoot. Frequently, the reason for the trouble falls under a couple of easy diagnoses.Here are some tips for identifying the most common reasons for one way audio, and how to fix them before they impact your ability to communicate with the outside world. Sep 07, 2019 · How to Resolve No Sound on Windows Computer. This wikiHow teaches you how to solve some common issues that result in no sound output on Windows computers. Keep in mind that your computer's issue might be too complicated to diagnose and fix
According Cisco audio may be restored if you put call on hold on cisco phone, but usually it is hard to explain to other side, especially when he doesn't hear you. We may say that it is phone's bug, but from other side it is not a very true behavior from asterisk's side to did such trick with the timestamp which is used for jitter calculation.
So, I have latest Asterisk 13.2, latest Crome (with Firefox - same problem) and sip.js (also tried with sipml5) and local network - no nat or firewall. The problem: if call is answered immediately - everything works fine. But if there are some delay in answer (say, 10 seconds) - no audio in both directions.
I have just upgraded an Asterisk 1.6.2-9 to Asterisk 1.8.13-1 (Debian distribution) and started to notice a problem with some peers : calls drop after 6-7 seconds and I have no audio. I have a lot of peers registered and experiencing the problem only with 2-3 of them. With Asterisk 1.6.2-9, same configuration everything was working fine.
Asterisk (Ver. 10.12.0) Ports forwarded: 4569 UDP 5060-5082 UDP 10001:11001 UDP faxdetect=yes vmexten=*97 context=from-sip-external callerid=Unknown notifyringing=yes notifyhold=yes tos_sip=cs3 tos_audio=ef tos_video=af41 alwaysauthreject=yes useragent=FPBX-2.10.1(10.12.0) disallow=all allow=ulaw callevents=no bindport=5060 jbenable=no defaultexpiry=120 maxexpiry=3600 minexpiry=60 allowguest Just logged into an older asterisk (1.8)and changed the progressinband to no and lost audio on a forwarded call to a cell phone. On another asterisk (13) system i have 6 did's pointing to 6 exts that have followme activated and progressinband did not work, had to play an announcement to get audio. Apr 27, 2018 · Asterisk supports a variety of audio and video media. Asterisk provides CODEC modules to facilitate encoding and decoding of audio streams. Additionally file format modules are provided to handle writing to and reading from the file-system. The tables on this page describe what capabilities Asterisk supports and specific details for each format. A no-way / no audio call is when you have a call between 2 phones (internal-internal or internal-external), and none of them can hear each other. How do I fix it? Before you start thinking about fixing that, you need to understand what is going on, how does it works, and what causes this problem. So, I have latest Asterisk 13.2, latest Crome (with Firefox - same problem) and sip.js (also tried with sipml5) and local network - no nat or firewall. The problem: if call is answered immediately - everything works fine. But if there are some delay in answer (say, 10 seconds) - no audio in both directions. Here is a full SIP trace obtained while calling from my cell phone (06611#####) to a SDA hosted by OVH (01850#####) and attached to my main OVH SIP line (00332309#####). The purpose of the dialplan is to transfer the call to an external line (09508#####). The call is actually transferred but there is no audio. Im not getting audio from WebRTC to WebRTC clients. I work in a LAN environment. Situation - Call from JSSIP to JSSIP (same client) with a standalone Asterisk server. Problem. There is no audio at all when doing a call from 6001(JSSIP) to 6002(JSSIP). After a while some RTP packets are getting send, but not received.