search for: tromboning

...all 6 PRI cards in the Index. (2 for termination to PSTN, 2 for outbound calls from Index to Asterisk and 2 for calls back from Asterisk to Index). Is this feasible? Can anyone offer some tips/advice based on their experience? I've had a look around and I can't find anything relating to tromboning or anti-tromboning so I suppose each call will have to take the path of: PSTN | Index | Asterisk | Index | Agent Regards Matt

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

[thread moved from -dev due to non-dev content] At 6:40 PM +0200 on 8/13/05, Andreas Sikkema wrote: >On Sat, 2005-08-13 at 12:44 +0800, Steve Underwood wrote: > >> He doesn't seem to really understand the strengths and weaknesses of >> IAX. IAX has drawbacks, but none of the problems he lists actually exist. > >OK, I'll bite ;-) > >How would IAX2 solve

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

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

...gt;> you are on and flashing it won't help. > > >> What you would need to do is get the other leg of the call to make the > >> flash. > > >> Of course if you where on a PRI link, you could do "hairpinning", "ect" > >> or "tromboning" and get the call taken back by the PSTN and transferred > >> to the new number. > >> -- > >> Steven Critchfield <critch@basesys.com> -- Steven Critchfield <critch@basesys.com>

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

...through Asterisk it 'rings' the extension test at foo.bar.com as if it were a standard SIP phone. That is stage 1. Stage 2 is for the media server handling test at foo.bar.com to be able to forward the call onto another SIP phone, allowing it to drop the call completely and the call tromboning or bridging to happen in the Asterisk PBX and not take up 'lines' on the media server. (Eventually both SIP ends may become PSTN's). I'm having trouble deducing how to do stage 1. In asteriskNow I don't see a way to add a calling rule for this. I tried adding a service pr...

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