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@_XY_
@_XY_ Жыл бұрын
My employer uses audio code
@voipsupply
@voipsupply Жыл бұрын
Great choice!
@gigawatt70
@gigawatt70 Жыл бұрын
This video was a lifesaver - thank you for posting it!
@voipsupply
@voipsupply Жыл бұрын
You're very welcome! Glad we can help!
@reneaguilar9351
@reneaguilar9351 Жыл бұрын
Can you do a tutorial mp114 audiocodes works with Vonage vdv21?
@voipsupply
@voipsupply Жыл бұрын
Hey, Rene! Thanks for the suggestion! We'll see what we can put together on our end. Stay tuned!
@reneaguilar9351
@reneaguilar9351 Жыл бұрын
can you do a tutorial mp114 audiocodes works with switchvox?
@voipsupply
@voipsupply Жыл бұрын
Hey, Rene! Thanks for the suggestion! We'll see what we can put together on our end. Stay tuned!
@adonaros
@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
@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
@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
@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
@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.
@KillerDonuts27
@KillerDonuts27 2 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
@adonaros Жыл бұрын
loop current disconnect setting.
@Panko-Evgeniy
@Panko-Evgeniy 2 жыл бұрын
А что если позвонить с внешнего номера через транк?
@voipsupply
@voipsupply 2 жыл бұрын
найди меня, следуй за мной, работает как с внешними, так и с внутренними вызовами
@Panko-Evgeniy
@Panko-Evgeniy 2 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.
@voipsupply
@voipsupply 2 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/
@voipsupply
@voipsupply 2 жыл бұрын
Похоже на проблему с конфигурацией. Вы можете обратиться в Sangoma за помощью в решении этой проблемы. www.sangoma.com/support/
@Panko-Evgeniy
@Panko-Evgeniy 2 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?
@cindesetiawan9175
@cindesetiawan9175 2 жыл бұрын
Thank You..., it work with 3CX Server
@seamuswarren
@seamuswarren 3 жыл бұрын
The phone network goes down in a blackout, which did not happen over the copper network.
@voipsupply
@voipsupply 3 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!
@alaincaulier2075
@alaincaulier2075 3 жыл бұрын
Hello, Bonjour de France. J'ai un souci, comment passer les deux communication du MP112 en meme temps vers le ipbx?
@voipsupply
@voipsupply 3 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.
@alaincaulier2075
@alaincaulier2075 3 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+
@voipsupply
@voipsupply 3 жыл бұрын
Hi Alain! Thanks for the comment! Unfortunately, we do not have a video for this at the moment.
@shubhamsoni2274
@shubhamsoni2274 3 жыл бұрын
A one minute video cleared the doubts of almost one week great job mate 👍
@mikec-hamilton
@mikec-hamilton 4 жыл бұрын
Just came for info on the Adtran unit. Well done video.
@Daniel-qo9uv
@Daniel-qo9uv 4 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?
@voipsupply
@voipsupply 4 жыл бұрын
Hi Daniel, I believe we used a preexisting trunk, but I will get the answer for you!
@arielmazuz830
@arielmazuz830 4 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?
@enriquegabriel7708
@enriquegabriel7708 4 жыл бұрын
Amazing!
@rolandomagallonmagallon2015
@rolandomagallonmagallon2015 4 жыл бұрын
Good job
@fatgirlboy9341
@fatgirlboy9341 4 жыл бұрын
no pros and cons. Stupid vid.
@ronaldnexus2378
@ronaldnexus2378 4 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!
@voipsupply
@voipsupply 4 жыл бұрын
Glad we could help, Ronald!
@nusrathsulthana8610
@nusrathsulthana8610 4 жыл бұрын
Good content better avoid music
@kristiallen3763
@kristiallen3763 5 жыл бұрын
IP telephony is one of the finest technology to make calls across the world. callhippo.com/cloud-based-virtual-phone-system
@VoIPPortland
@VoIPPortland 5 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 :(.
@gamutnyc4675
@gamutnyc4675 4 жыл бұрын
You got that right!
@scanalot7582
@scanalot7582 5 жыл бұрын
does not work for me I did it any other ideas?
@voipsupply
@voipsupply 5 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!
@saleysam6
@saleysam6 5 жыл бұрын
how i can reset by code or icon ? not on button from backend
@MrShish0k
@MrShish0k 5 жыл бұрын
Many thanks! Very quickly adjusted everything. Competent manual
@magpieenterprise6781
@magpieenterprise6781 5 жыл бұрын
Thank you
@MrBossZone
@MrBossZone 5 жыл бұрын
what if I can't get it to work on Cisco call manager? MT 505 says, "NO IP".
@voipsupply
@voipsupply 5 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.
@hariprasad984
@hariprasad984 6 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
@voipsupply
@voipsupply 5 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.
@hariprasad984
@hariprasad984 5 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.
@voipsupply
@voipsupply 5 жыл бұрын
@@hariprasad984 No problem! We're glad you got it figured out!
@mayushiideki
@mayushiideki 6 жыл бұрын
Try being more fluid with your speech.
@michaellengenfelder8928
@michaellengenfelder8928 6 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.
@voipsupply
@voipsupply 6 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
@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_Ka
@Yves_Ka 6 жыл бұрын
OMG he is soooooooooooo cute
@santiagodaviddelgadocarcac6272
@santiagodaviddelgadocarcac6272 5 жыл бұрын
Dude you gay
@tularem
@tularem 4 жыл бұрын
@@santiagodaviddelgadocarcac6272 lmao
@iOSINT
@iOSINT 6 жыл бұрын
убогое устройство, просто замучался менять на нём IP
@gauravzerogravity
@gauravzerogravity 6 жыл бұрын
opening music is a great Music on hold
@mohammedjahangir5754
@mohammedjahangir5754 6 жыл бұрын
Grandstream product is waste i purchased this one no support nothing KSA every time we have find someone for support.?
@matthawaii
@matthawaii 6 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.
@voipsupply
@voipsupply 6 жыл бұрын
And now Plantronics owns Polycom!
@Bloodyslayer73
@Bloodyslayer73 7 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.
@voipsupply
@voipsupply 7 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!
@MyEconoBlog
@MyEconoBlog 7 жыл бұрын
I use Sangoma with FreePBX at work. We love it! #Sangoma12
@siptellnet
@siptellnet 7 жыл бұрын
The feature of Trunk Groups is really a saving money opportunity
@hmbrasil2
@hmbrasil2 7 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!
@takondwachitheka7283
@takondwachitheka7283 6 жыл бұрын
hi, thanks for the tutorial. Do you have another tutorial on setup of mp 124 ?, please help
@voipsupply
@voipsupply 6 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
@billrathbone7572
@billrathbone7572 7 жыл бұрын
Wish you would show the wall mount setup (including the cat5 cable connections)
@brandonjones9635
@brandonjones9635 7 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.
@richmanruler7457
@richmanruler7457 7 жыл бұрын
what is the default ip for DP750 and DP720?
@davidread5550
@davidread5550 7 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?
@voipsupply
@voipsupply 7 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.
@kensrona
@kensrona 7 жыл бұрын
it would be nice if you could add in NFS authentication and remote door open.
@supersmart671
@supersmart671 7 жыл бұрын
Good job!!
@Apkabhai-Cricketer
@Apkabhai-Cricketer 7 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.
@voipsupply
@voipsupply 7 жыл бұрын
Hi Hammad Please ensure you have added the APs to all the zones (SSIDs/VLANs).
@HJBProductionsLLC
@HJBProductionsLLC 7 жыл бұрын
What is the PC connector for?
@voipsupply
@voipsupply 7 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).
7 жыл бұрын
great video
@napalmloveskids
@napalmloveskids 7 жыл бұрын
This dude sounds like an NPR radio host