similar to: Hangup detection and trombining

A new recording is now up on pan.zipcon.net and also gas.zipcon.net. Here is the Readme: The Virtuoso Trombonist Dennis Smith plays with W.W.S.S Wind Ensemble Wm Cole, Conductor, and with Martha Goldstein, organ. At ogg q=7. <p>In the years befor world war 2, Sunday afternoons might be spent in the park listening to the Municipal band. Selections 1 and 2 would be this kind of

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

Yes, After call transfer,I don't want to be media go through Asterisk. Is it possible ? Thanks, Karun. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Dameon D. Welch-Abernathy Sent: Thursday, August 12, 2004 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Blind Call Transfer using

Since 2008, Microsoft (formerly Revolution Analytics) staff and guests have written about R at the Revolutions blog (http://blog.revolutionanalytics.com) and every month I post a summary of articles from the previous month of particular interest to readers of r-help. In case you missed them, here are some articles related to R from the month of November: R 3.4.3 "Kite Eating Tree" has

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hello, If I connect an Asterisk 1.6 to a PBX via Q.SIG and A (on the PBX) calls B (a SIP phone on Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. Is it possible to configure Asterisk so that it does the Q.SIG ?Path Replacement Feature? ? The Q.SIG "Path Replacement Feature" requires the following: After both legs of the

Hey Folks, Is it possible to transfer an incoming call back out without a "trombone" effect. For instance; Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the caller selects an option --> asterisk transfers the call to my cell phone via broadvoice and removes itself from the equation so I end up with... Caller --> Broadvoice --> Cell Phone Vs.

I have been successful in installing and running Finale 2000 with Wine. The only - major - problem is that the music symbols are replaced by squares (like the default symbol caracter), making the music unreadable! I thought that importing the Maestro fonts with DrakeFont (Mandrake 8.0) and making them available to the system would help, but it doesn't. I have tried to add : Alias0

I have successfully installed and launched Sibelius (music typesetting software) on linux (with codeweavers preview 4, w/ a fake_windows install). It works fairly well, and would probably deserve a 4 on the app database. Except for the fonts, which kind of defies the purpose of a typesetting software! Here are the symptoms and some log activities : * No musical symbols are displayed (except

On Thursday, February 20, 2020 10:54:02 AM CST Fred Smith wrote: > Hi! > > totally OT... > > Hoping there is a mailing list or wiki (or other help forum) > for GWC, but haven't found one yet. > > I'm working on converting a bunch of my LPs to CDs, and am using > GWC (Gnome Wave Cleaner, or GTK Wave Cleaner) to clean up the noise. > It works great, but I

Check to see if you can get a IOS code leverl that supports SIP on the 6500. then maybe you can use your E1 card directly. you can also get a SIP version of the code for the 7960's etc Dave >>> jwi@weball.csis.dk 6/28/2003 2:56:12 PM >>> Hi Chris I've done a lot of things with Cisco AVVID solutions in the past. > CallManager).....am I right in saying that Cisco

Hi Everyone, I am trying to find the issue of dropped calls in the middle of the conversation. The system is Elastix. Anyway to know which party hangup the channel in case of Asterisk 1.4 and Sangoma analogue cards? (this is not PRI) Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and

Hi, I've set an Asterisk 11 box with a TDM400 board and Dahdi 2.7.0.1. I've connected an FXS port to an FXO one and issued a couple of channel originate command to measure the duration Asterisk/Dahdi needs to detect a dahdi call is coming in. Basically, using EPOCH variable, I'm reading a 2 or 3s duration with the followinf AEL2 dialplan: context remote { s => { if

Hi. Has anyone experienced hangup detection problems with the VoiceMail2 app? I have a console phone on the FXS port. When I call a SIP phone, and get its voicemail greeting, I can enter the VoiceMail2 app, leave a message, and then hit # to stop message recording. Recording does stop, but the channel stays up inside the VoiceMail2 app (as shown by a "show channels" command) for about

Many thanks to Martin Pycko and Mark Spencer. Mark's suggestion below was correct: "Maybe it's stuck trying to send the e-mail notification. If you take the e-mail address out of /etc/asterisk/voicemail.conf does that speed it up?" Indeed it did! The problem turned out to be a 60second delay while invoking mail, caused by a mis-configuration of my hostname and

Hello, I've got the following configuration: 2 X101Ps Asterisk built with BUSYDETECT_MARTIN busydetect=yes busycount=10 callprogress=yes signalling = fxs_ks With this setup, the best I can do is get voicemail with 17 to 19 seconds of silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has anyone had any success with this? It seems that hangups are indeed detected,

Hi, We have a system that recorded voicemail for about an hour after the caller hungup. I'm going to put a timeout on it but is there anything to look for that can help prevent this? The system is running on a telenet line in Belgium. The answer dialplan I used was: [macro-stddial] exten => s,1,Answer exten => s,2,Playback(transfer) exten => s,3,Dial(${ARG2},60) exten =>

Hi I'm having a problem with the X100P, where it doesn't detect the caller hanging up before it has answered. The result is that if the caller hangs up just before asterisk answers the line, it still picks up the line and records dialtone in my voice mail. Is there any way to fix this? I'm in the UK, and I use opermode=1 when loading wcfxo. Cheers Rob

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I