I picked up a Sangoma AFT-Remora A200 with 4 FXO ports from eBay for a home project I want to try. It did not come with any cables. The specification here states that the interface is "4 x 4-pin RJ11/4 narrow jack". A standard RJ11 jack does not fit. Does anyone know what is required here if I want to make my own cable? Is it RJ10 at one end and RJ11 at the other perhaps? I have seen the CABL-629 which may be the cable to use, but there are no details on its connectors -- and while they are cheap to purchase the shipping costs are ridiculous. If I don't make my own does anyone know of other o ..read more

I have one queue which servs several online shops, with the same dynamic agents. Trying to find a solution how to play a different sound to client for saying "we are too busy now..." specific to each shop after some waiting time. According to the docs, we can set from dialplan 'announceoverride' parameter for Queue() app, but can't set periodic-announce or queue-callswaiting file. Any suggestions? submitted by /u/blessend0r [visit reddit] [comments ..read more

I am just diving into the VOIP world, I have FreePBX working with a test environment. I am testing with one co-worker and they use an Intel NUC with built in microphone. When we use teams our communication is crystal clear. When I use a softphone like Zoiper or MicroSIP their microphone is very bad. I can barely hear them. They can hear me fine though with my headset. I was trying to avoid having this user require a headset and was hoping to use their NUC as it does work well. I played with all the microphone settings but cant get anywhere. I am using just UDP communication, any reason why th ..read more

I'm looking for information on setting up Asterisk as a Telephony Application Server. For reference, this is what I'm looking to use Asterisk as: https://www.dialogic.com/glossary/telephony-application-server-tas https://www.mobius-software.com/documentation/Mobius+TAS https://www.ericsson.com/en/portfolio/cloud-software-and-services/cloud-core/communication-services--udm/cloud-ims/multimedia-telephony-application-server We are currently using Metaswitch Rhino/Sentinel but Microsoft decided to pull it from the market. We are now looking for a carrier-grade replacement that we can support in ..read more

Hi everyone, We're in the process of setting up a small call center for our company (2 people). We have a VOIP number with SIP trunk credentials, and we've installed Asterisk and FreePBX on an Ubuntu server. We're looking for guidance on how to configure the SIP trunk and set up the call center so that both operators can access the VOIP line. Here's what we need: When a customer calls our number, they should be placed on hold with some music. The call should be forwarded to both operators. The first operator who answers will take the call. Also, we're not sure what these priority things mea ..read more

Hi everyone, I'm currently working on an AI call center project and need some advice regarding the best approach to handle incoming calls and perform automated calls using a PBX system. My client has a PBX, and I'm trying to decide between using a SIP trunk or PJSIP (specifically psua2) to handle up to 100 simultaneous calls. Here are the specifics: The system needs to manage both incoming calls and perform automated outbound calls. The target is to handle up to 100 simultaneous calls efficiently. I have some experience with Asterisk but would appreciate insights on the following points: S ..read more

Hello, I’m lost lol. I am migrating servers and while doing this I’d like to upgrade my asterisk instance. Sip.conf was way easier than the new pjsip and I am lost. I use twilio for my number and trunk provider. Can someone show me an example of a pjsip and extensions config with twilio for some guidance? submitted by /u/Xorfora [visit reddit] [comments ..read more

I don’t know if this is an older question because I’m not familiar with sip trunk and asterisk that much. Do sip trunk providers still allow you to set your own caller id for outbound calls? I heard that there are new regulations but I saw some older posts that it’s possible. If it is possible, could someone be kind enough to give me a list of providers that do it? submitted by /u/N0tA1dan [visit reddit] [comments ..read more

Do anyone know of a way to automatically normalize the audio volumes for calls going through Asterisk? I have been searching but not finding anything. submitted by /u/JosefAndMichael [visit reddit] [comments ..read more

Hi all, after adding a couple of endpoints to pjsip.conf and saving it, asterisk is no longer listening on port 5060. How can I restore connectivity? Thanks. submitted by /u/Outside_Simple_3710 [visit reddit] [comments ..read more