This video was a lifesaver - thank you for posting it!
@voipsupply Жыл бұрын
You're very welcome! Glad we can help!
@reneaguilar9351 Жыл бұрын
Can you do a tutorial mp114 audiocodes works with Vonage vdv21?
@voipsupply Жыл бұрын
Hey, Rene! Thanks for the suggestion! We'll see what we can put together on our end. Stay tuned!
@reneaguilar9351 Жыл бұрын
can you do a tutorial mp114 audiocodes works with switchvox?
@voipsupply Жыл бұрын
Hey, Rene! Thanks for the suggestion! We'll see what we can put together on our end. Stay tuned!
@adonaros Жыл бұрын
Howdy people.. I would like to remind you to check a few more settings on this thing.. sip deffinitions > dtmf and supp > max digits in phone number ... , north america 11? (because you dont have a dialplan set, and im not going to go there - this should be obvious) media > general media settings > nat traversal - on! dtmf & supp > supplementary services - here is where you turn off call waiting and enable CALLER ID!! control network > proxy sets table - ENABLE PROXY KEEPALIVE - using options (for most people) keep alive time somewhere between 10-15 maybe up to 30 for most people gw > hunt group > endpoing phone number - under "phone number" if youre using a sip username that is not a number, it goes here. notes: coders and profiles > coders > do add 711U and maybe make that one first (if youre in north america) SIP Definitions > Advanced Parameters - current disconnet - you probably really really want this if youre in north america
@EdGreenberg Жыл бұрын
Marc, this was fabulous. For the massive number of configuration settings, and the assumption in the manual that you know what these terms mean, you have simplified the most basic and common use case to a five minute process. I am looking forward to doing my MP114 in one window while your video runs in another.
@adonaros Жыл бұрын
Dont forget to go to sip deffinitions > dtmf and supp > max digits in phone number.. (also i seem to have a problem with NAT on this thing..)
@naveengupta3273 Жыл бұрын
We have patton fxo and fxs gateway, we are facing issue while coming incoming calls phone auto disconnected after 3 rings, plz tell what issue is
@voipsupply Жыл бұрын
Hi Naveen, Check the Patton "SIP Profile" within the WebUI administration configuration and Ensure that "Early-Disconnect" is NOT enabled. Please submit your issue to [email protected] if you will need further remote assistance.
@KillerDonuts272 жыл бұрын
@UCBqkepXGghBZCkUXmvmWTaQ - I configured an MP-114 to work with Zoom but when I use it as a paging system with my Bogen UTI312 I get a busy tone after hangup for about 6 seconds - which can be heard over the paging system. Any ideas?
@adonaros Жыл бұрын
loop current disconnect setting.
@Panko-Evgeniy2 жыл бұрын
А что если позвонить с внешнего номера через транк?
@voipsupply2 жыл бұрын
найди меня, следуй за мной, работает как с внешними, так и с внутренними вызовами
@Panko-Evgeniy2 жыл бұрын
@@voipsupply Я тоже так думал. Но когда подключил GoIP к Астериску и захотел с него направить звонок на внешний номер SIP телефонии, звонок не состоялся. SIP телефония нам предоставляет 10 внутренних, своих номеров, парочку из которых я подвязал по транку в Астериск. Так вот по этим транкам звонить могу на оставшиеся номера, а этим же транком переадресовать на эти оставшиеся номера не выходит.
@voipsupply2 жыл бұрын
@@Panko-Evgeniy Sounds like a configuration issue. You may want to reach out to Sangoma for support to solve this issue. www.sangoma.com/support/
@voipsupply2 жыл бұрын
Похоже на проблему с конфигурацией. Вы можете обратиться в Sangoma за помощью в решении этой проблемы. www.sangoma.com/support/
@Panko-Evgeniy2 жыл бұрын
@@voipsupply Спасибо. Попробую. Сегодня зашёл в конфиг nano /etc/asterisk/followme.conf А он вообще пустой. Возможно конфигурация Find Me/Follow Me в другом каталоге или файле расположена?
@cindesetiawan91752 жыл бұрын
Thank You..., it work with 3CX Server
@seamuswarren3 жыл бұрын
The phone network goes down in a blackout, which did not happen over the copper network.
@voipsupply3 жыл бұрын
If the power goes out then one option for your business would be a generator. Now, if your analog system or SIP trunk goes down they can use a failover to route to another PBX or host. To protect the equipment you should use a UPS (Uninterruptible Power Supply). If the ITSP goes down, then they can proxy their phones to stay locally online. You can use call forwarding and/or a VoIP app to continue the calls using your mobile phone or even direct calls to voicemail!
@alaincaulier20753 жыл бұрын
Hello, Bonjour de France. J'ai un souci, comment passer les deux communication du MP112 en meme temps vers le ipbx?
@voipsupply3 жыл бұрын
Salut Alain, merci de nous avoir contactés! Vous auriez besoin d'une passerelle MP FXO / FXS pour faire le pont vers le RNIS pour ce faire. Hi Alain, thank you for reaching out to us! You would need a FXO/FXS MP Gateway to bridge to ISDN to accomplish that.
@alaincaulier20753 жыл бұрын
hello,Super, for me I am looking for the config MP112 with Alcatel oxo and confines Alcatel oxo to MP112. If you have the configue I thank you in advance? A+
@voipsupply3 жыл бұрын
Hi Alain! Thanks for the comment! Unfortunately, we do not have a video for this at the moment.
@shubhamsoni22743 жыл бұрын
A one minute video cleared the doubts of almost one week great job mate 👍
@mikec-hamilton4 жыл бұрын
Just came for info on the Adtran unit. Well done video.
@Daniel-qo9uv4 жыл бұрын
Hi have you built the trunking from scratch in this video or you have used a 3-party company to make the trunking?
@voipsupply4 жыл бұрын
Hi Daniel, I believe we used a preexisting trunk, but I will get the answer for you!
@arielmazuz8304 жыл бұрын
Hi, I configured my mp-114 and it works perfectly in the last year now i replaced my analog phone and according to the guide menu i need to change the analog input from pulses to tone... How can i do it? And also how can i control the speed of the ports?
@enriquegabriel77084 жыл бұрын
Amazing!
@rolandomagallonmagallon20154 жыл бұрын
Good job
@fatgirlboy93414 жыл бұрын
no pros and cons. Stupid vid.
@ronaldnexus23784 жыл бұрын
Just what I needed to setup my PBX analog line; was totally confused on analog ports. This KZbin made it easy to see, I got my FXO port setup in five mins! Thanks guys!
@voipsupply4 жыл бұрын
Glad we could help, Ronald!
@nusrathsulthana86104 жыл бұрын
Good content better avoid music
@kristiallen37635 жыл бұрын
IP telephony is one of the finest technology to make calls across the world. callhippo.com/cloud-based-virtual-phone-system
@VoIPPortland5 жыл бұрын
If you never allow the phone to auto load FW from Grandstream and pre-test every release before allowing updates, the 2170/2135 is great. STILL the best phone for the dollar. A much nicer choice than the newer line of GS phones (Yealink knockoff cheapies) - the 'carrier grade' phones :(.
@gamutnyc46754 жыл бұрын
You got that right!
@scanalot75825 жыл бұрын
does not work for me I did it any other ideas?
@voipsupply5 жыл бұрын
Sorry to hear this isn't working for you! If you have followed the instructions on FreePBX in the video and it's not working, then check with your IT department or Systems Administrator. If that is you, then feel free to send an email to [email protected]. This is our free email-based ticketing system. One of our technical support team members may be able to assist you!
@saleysam65 жыл бұрын
how i can reset by code or icon ? not on button from backend
@MrShish0k5 жыл бұрын
Many thanks! Very quickly adjusted everything. Competent manual
@magpieenterprise67815 жыл бұрын
Thank you
@MrBossZone5 жыл бұрын
what if I can't get it to work on Cisco call manager? MT 505 says, "NO IP".
@voipsupply5 жыл бұрын
Hi, Esteban - It sounds like a DHCP problem, but I'm also not certain it will work that well, or at all on call manager. I definitely wouldn't recommend it as an endpoint for call manager. I'd say that you need to make sure DHCP is working on your network first before attempting to configure it.
@hariprasad9846 жыл бұрын
Hi... I Configured audiocodes MP114, I make a call from analog to ip and I get 416 unsupported media with reason "SDP: DTMF payload for RFC 2833 is missing" from the PBX..how to solve this I have tried but of no use... please help
@voipsupply5 жыл бұрын
Hi, Hari. It's tricky to troubleshoot without doing any tinkering but I would suggest you make sure the DTMF settings are the same on both the PBX and the MP114.
@hariprasad9845 жыл бұрын
@@voipsupply Thanks for your reply... Actually the gate name and gateway address fields were same as the pbx domain name, hence the call was rejected with 415 unsupported (dtmf) msg ...may be my PBX is sending a incorrect reply. Now issue is solved after changing the gateway name. Thanku very much.
@voipsupply5 жыл бұрын
@@hariprasad984 No problem! We're glad you got it figured out!
@mayushiideki6 жыл бұрын
Try being more fluid with your speech.
@michaellengenfelder89286 жыл бұрын
Dear Mr. Spehalski, thanks for your tutorial. I really can´t expect what 'Renegade' means. I try to register to a AVM Fritzbox and I can´t put in something appropriate to 'Registrar Name'. The MP ist reloading the page immediatelly when I come over a view keys. (e.g. 192.16 -> reloading automatically). What is the best practice? Thanks, w/best regards.
@voipsupply6 жыл бұрын
Hi Michael, "Registrar name" is a friendly name, a.k.a a label for internal reference. Without being able to test, there might be some input validation on the MP112 that is rejecting what you're trying to enter. You can use any name you like as long as the MP112 allows it. I'm not sure what you mean by "the MP isn't reloading the page". If there's some strange behavior with the web user interface, you should try a different browser. If that doesn't help, you can provide a step by step of what you're doing to see the behavior and we can have a better idea of what they're trying to accomplish.
@EdGreenberg Жыл бұрын
There is something odd about the MP1xx devices. Certain alpha keys cause the page to reload. For instance, I tried to put in 'asterisk' and it reloaded on the 'r'. It's said that browser choice may alleviate some of this. So far, I resolved the problem by entering MySwitch which did not have an 'r' in it. A posting I read referred to a character other than 'r', so your mileage may vary.
@Yves_Ka6 жыл бұрын
OMG he is soooooooooooo cute
@santiagodaviddelgadocarcac62725 жыл бұрын
Dude you gay
@tularem4 жыл бұрын
@@santiagodaviddelgadocarcac6272 lmao
@iOSINT6 жыл бұрын
убогое устройство, просто замучался менять на нём IP
@gauravzerogravity6 жыл бұрын
opening music is a great Music on hold
@mohammedjahangir57546 жыл бұрын
Grandstream product is waste i purchased this one no support nothing KSA every time we have find someone for support.?
@matthawaii6 жыл бұрын
I never cared much for Obihai's desk sets. Polycom didn't need to buy them for their SIP stack. Polycom must have thought it easier to acquire Obihai's ATA products then build their own. I have expected Polycom to purchase/develop a PBX product like Grandstream did.
@voipsupply6 жыл бұрын
And now Plantronics owns Polycom!
@Bloodyslayer737 жыл бұрын
Hi Marc. I am trying to configure an Adtran 908 3rd Gen to use TLS. Any advise/tutorial? I am having issues uploading CA certificate.
@voipsupply7 жыл бұрын
HI Bloodslayer, You upload the CA certificate when you create a new CA Profile. To do that, navigate to: Data -> VPN -> Certificates, then create a profile. Once a profile is created, you can paste in the CA certificate and use it for the TLS configuration. We hope this helps!
@MyEconoBlog7 жыл бұрын
I use Sangoma with FreePBX at work. We love it! #Sangoma12
@siptellnet7 жыл бұрын
The feature of Trunk Groups is really a saving money opportunity
@hmbrasil27 жыл бұрын
Obrigado pelo vídeo, ajudou muito! Porém, tivemos muita dificuldade pois a conta SIP não registrava dejeito nenhum em nosso sistema (OpenSIPS), e com muito esforço só funcionou depois que preenchemos também o campo "Gateway registration name" com o IP do servidor. Fica a dica!
@takondwachitheka72836 жыл бұрын
hi, thanks for the tutorial. Do you have another tutorial on setup of mp 124 ?, please help
@voipsupply6 жыл бұрын
Hi Takondwa! We do not have a video for the MP124 but please feel free to put in a ticket to our support team! voipsupply.zendesk.com/hc/en-us/requests/new
@billrathbone75727 жыл бұрын
Wish you would show the wall mount setup (including the cat5 cable connections)
@brandonjones96357 жыл бұрын
Very interesting product for those of us who wish to go the VoIP route. However, I figure unboxing is the easy part. For example, it would be interesting to watch a video showing somebody setting up the base unit plus three or four handsets right out of the box to the point that they are actually operating in a home environment.
@richmanruler74577 жыл бұрын
what is the default ip for DP750 and DP720?
@davidread55507 жыл бұрын
Hey Marc, do you have any tutorials on how to configure the fxo 0/0 to talk with the freepbx so that I can have a backup local line in case my PRI goes down?
@voipsupply7 жыл бұрын
Hi David, and thanks for the comment! We currently do not have a video for this scenario, but we will add it to our list for the future.
@kensrona7 жыл бұрын
it would be nice if you could add in NFS authentication and remote door open.
@supersmart6717 жыл бұрын
Good job!!
@Apkabhai-Cricketer7 жыл бұрын
i saw ur tutorial i got a question. i had already configured the Access point that is very simple but still i am not able to transmit the SSID. i cant see the WIFI SSID on my network.
@voipsupply7 жыл бұрын
Hi Hammad Please ensure you have added the APs to all the zones (SSIDs/VLANs).
@HJBProductionsLLC7 жыл бұрын
What is the PC connector for?
@voipsupply7 жыл бұрын
Hi HJB Productions! The 'PC connector' will connect your PC to the phone, this will allow you to share a single network connection for both devices (phone and PC).