No audio on Asterisk SIP call. Ask Question Asked 9 years, 2 months ago. Active 3 months ago. Viewed 30k times 8. I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not hear one
Elastix - No sound to external calls from sip phones connected through NAT. Ask Question Asked 3 years, (2 way audio) using 3 digit extension numbers. Apr 19, 2019 · Yes/No: Whether to enable scheduled paging. You can optionally set schedules for any page group. You can optionally set schedules for any page group. The system will automatically call the page group at the selected dates and times, play the recording that you have chosen in the Page Announcement section above, and then hang up the call. Once you've set up your queues and started taking calls, you should also take a look at OrderlyQ, which is an add-on for standard Asterisk queues that allows your Callers to hang up and call back later without losing their place in the queue, resulting in substantial increases in Caller satisfaction and retention, and substantial savings for Call Center operators. Jul 14, 2013 · I was using Elastix 2.4 and had a very similar problem. Two LANs - one with Elastix in datacenter, two at remote offices. IPSec LAN-LAN VPN between them. If using the LAN-LAN VPN, I would get no audio. However, if the softphone or phone would VPN to the datacenter, then connect, everything was good. There is no way that the switching fabric is saturated, just not a realistic concern. QoS might be needed on the router but that would be exclusively for the audio that is sent, not received. So people calling into the office might hear things better, but it does not affect what people hear in the office itself. Elastix SIP Firewall is a frontier device, designed to be placed alongside a VoIP IP PBX in order to add an additional security layer. How does it work? It blocks specific IPs or countries, protecting your PBX against potential intruders trying to get in with usernames and passwords. No it is a P3 dedicated only to Elastix (and only the using PBX function) and it is not a complex setup. Also I have sometimes encountered quite a bit of jitter as well as the occasional call with big latency, a delay of about 5 seconds sometimes.
Hi, after replacing (an old) Freebpx installation with 13, the remote extensions are able to register, intitiate calls, but there is no audio. During the call setup I can see a 401 error, and after a few seconds the line is dropped because no response from the external extension. We are running a NAT setup, no SIP ALG, same NAT setting as the old freepbx system. I'm sure it's related to
The general problem with the no audio on remote extensions with asterisk is where you put the sip nat settings. Most people put it in the sip_nat.conf file but as these settings would be added at the very end of your SIP configuration it wont work. Some machines it works and a lot it doesn't. I have exactly the same problem here - no audio on call forward to mobile/landline phone. The only solution I have found is to tick the 'confirm calls' in the follow me settings for the extension with the call forwarding. Then the audio is connected only when I press 1 on the mobile after the call is connected. No audio on Asterisk SIP call. Ask Question Asked 9 years, 2 months ago. Active 3 months ago. Viewed 30k times 8. I almost managed to init a 2 sided call (click to call): 1st to my office and the 2nd to my cell using Michal Niklas answer (thanks Michal) on Asterisk click to call. The major ISSUE is that the 2 call participants can not hear one
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Elastic Audio is one of those things that can often get us into more trouble than we bargained for. Recently I was asked to bolster a sequenced drum line with some real life drums, here's the easy Jan 22, 2013 · Adding/Enabling G.722 HD Audio Codec on Elastix (or FreePBX) On January 22, 2013, in Technology , by Mike Waldron I recently have had prospective clients ask about HD voice/G.722, so I figured I’d experiment and see if it’s supported in Elastix. Elastix is the first distribution to include an Open Source Call Center module with a predictive dialer. This module can be installed from the same web-based Elastix interface through a module loader.