search for: tromboned

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

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

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

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.

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

...2 solve trombones? > >-- >Andreas Sikkema Since this is a very vague accusation of protocol shortcoming, I'll answer in a very vague way: IAX2 has the ability to "native bridge" two endpoints together, even if the call was established by a third party. Media is not "tromboned" through the third party if it is possible for the two endpoints to communicate directly, and it is administratively permitted. Please read the (not quite completed) IAX2 specs. http://www.cornfed.com/iax.pdf JT

...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 call are setup and Asterisk has a successful "tromboned" or bridged call ... Asterisk sees both 'B' channels (from trombone) are on same PRI/technology and initiates "Path Replacement" events a. Asterisk sends "Transfer Complete" messages to both call legs b. QSIG Switch sends "Purpose" message on one o...

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

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

Hi! If I install a CAPI-compatible ISDN-card in my server, will that: a) enable me to connect that server to the public phone net b) allow me to connect an ISDN phone to the server and use it as a SIP-phone c) all of the above? Regards, Evert

Hey.. Your saying I can not use flash with ISDN ? What options to I have to transfer a call directly ? ( So I have a free line afterwords) >> What interface are you using? ZapBRI? if so you might be able to do the >> hairpinning if it is supported. Im not using any interface.. But if you know how to do that, let me know and I install that interface. Thx for your answer :) Gr?sse

Hi All, I hate to post yet another "bloody hangup detection issue" on the list, but I have been pulling my hair out no end of late with a hangup detection issue on 1 system (have a few others out there with TDM400's and no issue but this one is causing a real headache) The scenario is - system with TDM04B, a call comes in on a analogue line, rings internally and then diverts to a

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

Hi, I am learning Asterisk for a small project. At this stage I have an AsteriskNOW system running locally. I can call SIP phone to SIP phone fine, the operator and voice mail work fine, except some stuttering (probably caused by it running in MS-VPC) What I need to figure out is... I have an automated response telephony server (Voice Media Server) available via a SIP URI like:

Hey Guys Im trying to forward a call with asterisk to a regular phone. Something like " I get a call on my regular phone, and he's trying to reach some buddy of mine.. then I tell him "wait a sec" and push "Flash" and get a other dialtone.. then I dial that other number then hangup the phone, so the one that called will be connected to where I dialed it to"...

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

I'd be grateful if anyone here could comment knowledgeably on an idea that I have had, as to whether it should be possible or not. Consider two Asterisk boxes, each with one or more E1s on EuroISDN. Each box has a different telephone number that hunts across all its E1 channels. In addition there is another number that hunts across all the channels on all the boxes. A call comes in to one of

Greetings, This may be a bit arcane but does anyone know what the contents of a facility message should be for initiating a TBCT on an NI2 ISDN. I've been trying to get it to work on a DMS100 for the last four months to no avail. The message I am currently sending makes it to the switch but is returned with unknown message. Perhaps someone here has done it before and can help me out.

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 can't figure out how to deal with certain types of pressing flaws that create a thump every time around. Anyone