similar to: DTMF Error

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

I am having a problem trying to do inband DTMF passthru via asterisk. My setup: PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2) Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly) 12/10/04 and 01/17/05 (no difference) CAC ABII-T100P to/from analog lines to/from asterisk BTW, I have used a ABI and it works just like the ABII with asterisk. What I am seeing is: I make a call from a

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I

I had a working configuration whereby an incoming call on an ISDN line would be sent out on the second ISDN line, but since I updated to the latest version of Asterisk I get this error message: WARNING[311315]: File res_parking.c, Line 226 (ast_bridge_call): Bridge failed on channels CAPI[contr1/s]/0 and CAPI[contr1/01624619052]/1 The message comes up as soon as the outgoing call is answered

Does anyone have any idea what would cause distorted audio but ONLY for DTMF tones coming in over our analog lines. (The analog interfaces are X100P's). I have carefully adjusted the gains in the zapata.conf using a local test line after trying various settings with no gain or just random gain settings. RelaxDTMF has no effect. I set up a monitor command in my dial plan to capture

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

Check with your telco. We had the same problem on 1 of our PRI's, every day at 5:00 sharp, red alarm, with all calls cut off for 30 seconds exactly. Turns out the equipment at the CO was going into a test loop at that time because of a forgotten setting by a tech. Man, what a finger pointing exercise that was. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent:

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

Hello, We have 15 Soundpoint IP 500 phones and recently call quality has deteriorated. On some calls there is a static-buzzing of sorts that occurs when users talk. It can be picked up on SIP-SIP calls and SIP-ZAPTEL (Channel BANK<<-T100P). It basically sounds really weird whenever someone talks, it sounds like a bee buzzing or something. Very hard to explain. Also, there will be echo

I have used every asterisk 11.8.X version. Have not had an issue till 11.8.0 and 11.8.1 When I use ConfBridge I am attempting to put all participants in MUTE mode and just one talker... However, since 11.8.0 I am hearing feedback in the announcement like the channel is not really muted. I dropped back to 11.7.0 and I hear no feedback. Has something changed - or - am I not correctly setting up

Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and

Hi?Danny, Thank you for your prompt response. The way you are suggesting is great .?Infect?asterisk have its own functionality that if user presses *1 during meetme conferencing asterisk automatically unmute that user and user comes in talking mode.But it is not?fulfill my need. There is and issue that if 3-4 user presses *1 at the same time than how can i decide that who is asking the question

CentOS-6.6 Does anyone know of an application that would provide a 'mute' buttons for video and audio that could be used from a desktop panel? Ideally this would show the status (muted/cloaked or open mic/recording) of each. Presently, turning audio and video pick-ups off and on is rather cumbersome. And I often forget what state I have left them in. Which occasionally makes for some

Hello, About every 2 or 3 days I notice in the messages log file: Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 2: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 3: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 4: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on channel 5: Red Alarm Dec 17 08:39:27 WARNING[1220]: Detected alarm on

Hi I am using asterisk 1.6.0.5. I have one conference say 1234786 and in this conference 25 users are talking with each other.. In this 25 users, 5 is admin/marked and 20 are normal.. Admin user has rights to mute/unmute all user by executing action: meetmemuteall with meetme number. While executing MeetmeMuteAll action, this action will mute all 20 normal users but not admin.. This thing work

Hello list, I recently started testing the chan_mobile addon and after a successful installation and configuration I have a couple of problems that I can't fix without your help. I am using opensuse 11.1, asterisk 1.6.1 with bluez 4.22 (installed from rpm packages) and a Nokia N80 phone. Apparently all works fine except the DTMF. Seems impossible to catch DTMF when nothing (no song) is

Hi, Any suggestions to below problem? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jamie Heckford Sent: 17 October 2006 21:48 To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect callif not dialed quick enough? Hi List, Have an odd problem