วีดีโอ

www.patreon.com/posts/freeswitch-sip-111210546?Link&

So, are you saying the SignalWire connector is easier than setting up a SIP trunk? Is that right?

@omidmohajerani like so much easier. Also signalwire works so much better with FS than other platforms. I am integrating FS with didww. Not a nice experience. Thankfully I was able to do it with your help. Need to read more docs and automate the process with python3 module now. 😅 Tysm, great series of videos.

Thank you.

@@omidmohajerani I faced another issue when I was integrating another platform called DIDWW into freeswitch. Luckily I was able to solve it, but the issue was when I did incoming calls to from the DIDWW into the freeswitch, calls wouldn't go to the extension and context defined in the gateway/didww.xml, and instead go to the public context because in the sip profile the context was public. :| I explicitly needed to give the context of the extension, and only then it would try the extensions. Weird. Any links to docs for this? Or has freeswitch changed in the time this tutorial was made.

Appreciate your comment! It’s great to know you liked it.

Appreciate your comment! It’s great to know you liked it.

What is the problem ? Its quite straight forward . Set a sip carrier in vc to connect to yeastar and define inbound did to your ingroup or ivr

@omidmohajerani need configuration steps,

I do it before with goip and work but with yeaster gateway cant do inbound configure

@@alysayed7299 Unfortunately I dont have one

Hi , you can pm me in patreon as you are memebr and i can help to check it.

@omidmohajerani thank you Omid chat Got solve the issue.I'm requesting you to please make video on installing vicidial from scratch on any cloud provider through OS ubuntu 24 & Centos 9 Thanks.

@ Happy to hear your problem is resolved: goodluck.

Thank you for the info. yes. All my German friends told me that even its not German :) Of course, it seems it's improved a lot since then but still...

Thank you.

Hey, I have not used it much and for other languages . Nowadays i usually use signalwire or twilio . So i send the call via sip to them and then i use their apis …

@@omidmohajerani ok…for Italian you need to modify languages.js and provider.js the first const supported = [ “en-US”, “it-IT”, ]; the second static languages ​​= [ “en-US”, “it-IT”, ]; And const DEFAULT_LANGUAGE = “it-IT”;

@@simonepalomba4455 Thank you for the info and update

Hi , And what is error ?

Thank you

VoIP Monitor is not my product; I only introduced it in my videos. While both VoIP Monitor and Homer offer similar functions, there are some key differences. VoIP Monitor is a paid solution, although it offers a 30-day trial license for testing. Homer, on the other hand, is free and open-source but lacks some advanced features, such as RTP metrics like MOS. To access these metrics with Homer, you would need to upgrade to their commercial product, called HEPIC.

Found the issue. Weblish console doesn't work anymore for the final pfSense set-up but the Glish console presents the right options for set-up, so must use that! Great video btw and really helped - no-one else seems to have done a video on this strangely ;-)

thanks for sharing . Yes I just needed for one project and never used it in such a way anymore. Im glad it helped.

Same as debian . I have a video for installing freeswitch on debian here th-cam.com/video/wx83YnEtXVo/w-d-xo.html

To change the storage path for recordings, multiple configuration steps are usually required. To simplify this and avoid potential issues, a better approach is to sync the contents of the /var/spool/asterisk/monitorDONE/ folder to your target destination and then create a symbolic link for each file pointing to the new location. This can be automated using a shell script.

Hi. Im not sure if fusion pbx has a read feature for it bit you can define an application before sending the call out using ring_ready application. you can search for ring_ready application in freeswitch to get an idea on how it works.

Hi. Sorry I dont have a ready mangus billing installed and ready to work on it .

@@omidmohajerani please share your contact details

my email is just like my username here at gmail

@@omidmohajerani ok

th-cam.com/video/-glPg5-2O6s/w-d-xo.html

yes you need 9$ support plan.

Here is the link for the related video and download images. www.patreon.com/posts/installing-11-on-92601796?Link&

Yes it should work without problem

👍🙏

You can send me message in patreon id you have downloaded from there

@@voiceofrjbilal search for a local sip carrier in your country or use international ones like signalwire, twilio, etc.

You should install in a cloud provider instead of your laptop . Check this video on how to do it Installing VICIdial on DigitalOcean (Step by Step and the easy way) th-cam.com/video/tnBr9AQl0hs/w-d-xo.html

@@omidmohajerani Thank you for your time sir

@@omidmohajerani Can you guide me where i can get the voip route for my dialer and how can i attach to the dailer and also give me the link of the iso image for Do so i can use ? Thanks

I have videos on how to connect to some providers here : www.patreon.com/omidmohajerani .

@@omidmohajerani I tried 2 times for your official patreon package but i can't get access

you can use mysql command mysql -u root -p asterisk -e 'DELETE FROM vicidial_ip_list_entries (ip_list_id,ip_address) VALUES ("ViciWhite","192.168.2.100");' remove 192.168.2.100 with your ip address to remove

What do you want to achieve with a custom AMD?

What is your question? What error do you have that i can help with ?

Thanks, Karsten! 🙌 You rock the hardest, Boss! 🤘 Honestly, I learned about this and so much more all from you. You're always an inspiration!

Hello! It seems like there might be a bit of confusion. I haven't done any coding in this video. I think you're referring to the SSH shell I used to run commands. If you're using Windows, you can use PuTTY to SSH into your server.

@@omidmohajerani I dont know understand how i should add into my server. please guide me

@@omidmohajerani it says Keyboard-interactive authentication prompts from server:

@@AsterismGlobal You can google "How to Use PuTTY to Access Your Server"

Thanks and Goodluck

Without checking the logs is hard to say :) at least hard to say for me

Hi, if you have a question you can comment here . for Paid support and projects you can message me here: telegram.me/Omid_Mohajerani

Yes it’s possible to have the outbound agents answer incoming as well . Ofcourse not recommended when you have predictive dialer campaigns because you need to reserve some capacity.

No, I haven't created a specific installation guide for Docker. If you're looking to set up VICIBOX in Docker, it would require a custom installation from scratch, as there is no pre-built VICIBOX Docker image available.

Hi, No Im not working on carrier routes. Im not aware of a good one with that capacity.

Hi if you are one of my gold users in patreon you can send message me there and i will check www.patreon.com/omidmohajerani/membership

Do you want to route multiple DIDs to one IP PHONE ? Or you have multiple ip phones ? Do you have pbx like Freeswitch or asterisk installed and configured ?

I want multiple did numbers for outbound campaign​@@omidmohajerani

You can use multiple did in your trunk in your pbx

Multiple DIDs in one IP phone and configure to asterisk.

🙏🙏🙏

yes. you can install VICIdial from scratch on Rocky Linux 8 or overall RHEL family version 8. RHEL 9 default packages make it very difficult, especially the PHP not supported in current SVN code.

Hi, The latest VICIbox version is 11, but version 10 remains stable and reliable. However, it's recommended to use the latest version for better security and features. For 100 to 200 seats, a full cluster setup is required, regardless of the version used.

The Dispatcher module is designed to route calls to a specified SIP destination, whether it's directing calls to voicemail, a conference system, or any other SIP endpoint.

Its an old post. Thaks for informing its still working. Goodluck.

Thank you.

yes you need to work with dialplan. search about SpeechBackground and also there are multiple ways to do it Silence detector is part of the DSP: github.com/asterisk/asterisk/blob/master/include/asterisk/dsp.h Example implementation: github.com/asterisk/asterisk/blob/master/funcs/func_talkdetect.c The res_speech module: github.com/asterisk/asterisk/blob/master/res/res_speech.c The res_speech_aeap module: github.com/asterisk/asterisk/blob/master/res/res_speech_aeap.c it’s up to you to piece it all together

Hi, yes! It's actually an advantage if you're new to VICIdial installation, as even the partitioning is handled automatically.