similar to: AudioCodes Gateway and Asterisk

Hi, I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go

I'm currently looking to deploy an Asterisk server using an FXO media gateway to connect to the PSTN and was looking for any user experiences that may aid in selecting a gateway. Specifically i'm looking for a 4-port model under 500 dollars. Within this category exists: MediaTrix 1204 Grandstream GXW-4104 AudioCodes MP114 I've read over Voip-info.org regarding these products and

I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO->SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does anyone have a setup file they can share to help me work this out. Instructions or a link I can follow - thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090726/643e5b96/attachment.htm

Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid -------------- next part

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>

Hi List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid -------------- next part

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more "SIP Aware". Anyone know one those that are out there ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071126/eb28ce44/attachment.htm

Razza, I have a MP114 FXO/FXS that I have never got to work , even as an FXS, even though I have several other FXS's that work fine ie Linksys PAP2 etc.. would you put up your config? PDE

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Hi, I have a client that wants a phone system that will play sounds from a sleep machine. I tried using all different formats (GSM, WAV, WV49, MP3 etc.). Over SIP it was OK however with the PSTN it broke up from time to time. I assume this has to do with the fact that the PSTN is limited to 8khz. Is there something I am missing here or is this simply a limitation of the PSTN? Regards,

I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa (ATA -----> Asterisk ----> SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP

<RANT> Funny your concerned about copyrights and moral issues regarding the work of others. One question you may want to ask YOURSELF is: Why would I use as my email a copyrighted work followed by the name of the Company that owns the copyright??? asteriskdigum@yahoo.com, Come on!! Who are you trying to fool? Are you out for the fast buck, by having someone that thinks you work for

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

Hello, I'm helping a colleague (*) which has the following setup: ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>-- Audiocodes MP-112 --- <FXO/FXS> --- Fax machine My issue is the following : Audiocodes gateway reject INVITEs with 488 Not Acceptable Here It seems this gateway requires t38 settings to be present in SDP body in the very first INVITE. My