Question about AudioCodes Inc. AudioCodes Mediant 1000 1SP MS UC Compatible M1K-UC10 Router

Open Question

An INVITE is sent to the gateway and before the called PSTN party answers, the calling SIP station hangs up, this results in the Audiocodes gateway reecting the CANCEL

Posted by on

2 Suggested Answers

6ya6ya
  • 2 Answers

SOURCE: I have freestanding Series 8 dishwasher. Lately during the filling cycle water hammer is occurring. How can this be resolved

Hi,
a 6ya expert can help you resolve that issue over the phone in a minute or two.
best thing about this new service is that you are never placed on hold and get to talk to real repairmen in the US.
the service is completely free and covers almost anything you can think of (from cars to computers, handyman, and even drones).
click here to download the app (for users in the US for now) and get all the help you need.
goodluck!

Posted on Jan 02, 2017

  • 2 Answers

SOURCE: AUDIOCODES MP-118 FXO

Hi, I am using Audiocodes MP-118 FXO too
Man, check the Reverse Polarity.

1st.- Go to Protocol Management
2nd.-Go to Advance Parameter
3rd.- Go to General Parameter
4th.- Go to Enable Polarity Reversal and select ENABLE
5th.- BURN and RESET

You have FAS calls, because Reverse Polarity is disable.

Lucky

Posted on Nov 04, 2009

Add Your Answer

Uploading: 0%

my-video-file.mp4

Complete. Click "Add" to insert your video. Add

×

Loading...
Loading...

Related Questions:

1 Answer

How to reset MP 124


It's - literally - right in front of you.

There is a pinhole on the front of the device; to the left of the display where it says "Channels."


24160359-ey2bbmgvpvvjdxgr04vnsi4f-4-0.jpg

Apr 17, 2014 | AudioCodes Mp-124d Mp-124 Fxs Analog...

1 Answer

I would like to connect an audiocodes mp 114 fxs to an Avaya Communications manager. The problem is Call sets up but no audio is heard. I also can not get the gateway to register with the avaya aura Ses...


Are you able to get a Wireshark trace from a mirror port of incoming/outgoing traffic for that MP? that way SIP session and RTP (audio) session can be analyzed and issue pinpointed. If no audio is arriving into MP no audio is expected to go out, if audio IS being delivered from the MP to the network, MP is doing its job. If audio IS getting into MP from network but no audio goes out further debug should be made, also if no audio is being delivered from MP to the network.

Jul 29, 2010 | AudioCodes MediaPack MP-114 FXS

1 Answer

AudioCodes MP-11R/2FXS


The easiest way (if enabled) is to use the voice menu as if you would assign a new IP to the device.

  1. Connect a telephone to one of the FXS ports. Lift the handset and dial ***12345 (three stars followed by the digits 1, 2, 3, 4, 5).
  2. Wait for the 'configuration menu' voice prompt to be played.
  3. To change the IP address, press 1 followed by the pound key (#).
  • The current IP address of the gateway is played. Press # to change it or hang up if you don't want to change it.
  • Dial the new IP address; use the star (*) key instead of dots ("."), e.g., 192*168*0*4 and press # to finish.
  • Review the new IP address, and press 1 to save it.

Of course you can try the default IP of 10.1.10.10 if the device is set to it's default standards.

If not, you can try a network scanner like angryip (link) or network scanner (link) - this might take a long time, since the IP can be virtually anything.

If you have physical access to the device, you can also use a straight RS-232 cable to connect the AudioCodes to your PC and use HyperTerminal to connect (Baud Rate: 115,200 bps (MP-124), 9,600 bps (MP-11x), Data bits: 8, Parity: None, Stop bits: 1, Flow control: None)

If you need the manual (and it never hurts ;-) here is the link:

http://www.audiocodes.com/filehandler.ashx?fileid=36320

If you have any more questions, just come back here and leave a comment - I'm happy to help.

Oct 22, 2009 | AudioCodes MP-112R/2FXS/3AC/SIP-3/UC...

2 Answers

Disable SIP AL


connect to the modem through telnet and type
application unbind SIP 5060 saveall

Aug 28, 2009 | Computers & Internet

1 Answer

Well I'm on reception and when I put calls through sometimes the other person on the other line has to push the call to pick up.. It only happens every now again. Most of the time I put the caller on hold...


Save yourself some time and decide how you want to process calls. If you press the DSS button directly, the outside line will go on hold and the inside party is rung. When they answer, if you hang up, the outside held call is immediately connected. If the inside party hangs up first, the outside held call stays in hold.

I would use the procedure "may I tell him who's calling?" Announce the call to the inside party as "I have a call from so-and-so" If they accept, you hang up.

Now, if you have to locate the inside party, press HOLD, then dial the extensions, page, or use the DSS buttons to find them. Tell them they have a call holding on line such-and-such, and they would know that they have to select the blinking line.

Back in the day, I used to be the fastest PBX operator in the place, and I just installed them :-) The key was to send it to a station you knew wouldn't answer and it put it out in a holding pattern before it recalled. This was in the day when you could only hold 3 lines at a time, so it worked well when the board got swamped. Today, with colored lights and timed recall, you don't need to do that.

Carl

Jun 24, 2009 | Panasonic BTI KX-T7640BK DSS Console BLACK...

1 Answer

AUDIOCODES MP-118 FXO


Hi, I am using Audiocodes MP-118 FXO too
Man, check the Reverse Polarity.

1st.- Go to Protocol Management
2nd.-Go to Advance Parameter
3rd.- Go to General Parameter
4th.- Go to Enable Polarity Reversal and select ENABLE
5th.- BURN and RESET

You have FAS calls, because Reverse Polarity is disable.

Lucky

May 28, 2009 | AudioCodes MP118 Analog Modem

1 Answer

Conference call


To make a Conference Call
1. While on a call, press Cnf/Trn_____.
2. Dial a [DN] (or access an outside line and
dial an external telephone number).
3. Press Cnf/Trn after the called party answers.
All parties are conferenced together.
If you added an outside line to the call, press
Cnf/Trn again before hanging up to allow
the outside parties to continue talking. (If
you do not, the call is disconnected).
Note Some types of outside lines
(“unsupervised”) do not automatically
disconnect when conferenced parties
hang up. In this case, press one of the
flashing buttons to monitor the
conference.
If the parties are still on the line, press Cnf/Trn
+ Spkr, then hang up. When no one is
on the line, press Spkr to disconnect the
lines.

Mar 19, 2009 | Toshiba DKT-2020 Corded Phone

1 Answer

Audiocodes MP118 8 FXO early media to Freeswitch SIP extension


Hi, man.
There are many things to consider in this type of application.
when calling from PSTN to IP using analog lines from time to time you will have this type of event, I suggest working with H323 protocol, this will tone down a bit your problem.
If you want optimum performance, you should use a DID with the channels you need and address to your PBX SIP FreeSwitch. The DIDs are cheaper and are digital which eliminates random connectivity problems in the incoming call.
Once you have directed the DID you tell your server to transfer the call to the IVR using your internal dialing plan.

Lucky

Oct 06, 2008 | AudioCodes MP118 Analog Modem

Not finding what you are looking for?
AudioCodes Inc. AudioCodes Mediant 1000 1SP MS UC Compatible M1K-UC10 Router Logo

Related Topics:

24 people viewed this question

Ask a Question

Usually answered in minutes!

Top AudioCodes Computers & Internet Experts

yadayada
yadayada

Level 3 Expert

75822 Answers

The Knight
The Knight

Level 3 Expert

74136 Answers

Are you an AudioCodes Computer and Internet Expert? Answer questions, earn points and help others

Answer questions

Manuals & User Guides

Loading...