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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
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.
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 hardwaresupport@voipsupply.com if you will need further remote assistance.
@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?
@@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 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?
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!
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.
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+
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?
Just what I needed to setup my PBX analog line; was totally confused on analog ports. This RU-vid made it easy to see, I got my FXO port setup in five mins! Thanks guys!
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 :(.
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 hardwaresupport@voipsupply.com. This is our free email-based ticketing system. One of our technical support team members may be able to assist you!
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.
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
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.
@@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.
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.
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.
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.
