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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

What happens to the block layer if the block request queue gets very, very long? Imagine there are many DomUs and they are all working against VERY SLOW disks. I wish I understood the entire chain better, and I''m sorry to ask such a vague question. I understand this is a weird question, because the performance implications of this scenario are horrific. The reason I ask is that

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

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

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

This is release 0.9.55 of Wine, a free implementation of Windows on Unix. What's new in this release: - Direct3D improvements, including driver version emulation. - Beginnings of support for OLE objects in Richedit. - Several fixes to the animation control. - A bunch of fixes for regression test failures. - Lots of bug fixes. Because of lags created by using mirrors, this message

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

[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

We''re trying to deploy our system on a separate server and have run into nothing but trouble; specifically I get the error mentioned in the subject. The username/password combo is correct (I verified it via SQL+). Manually running script/server works fine in production or development mode. Heck, running script/console also works fine. It''s just that when it automatically

Greeings, I'm running rsync version 2.5.4 on a solaris 2.8 box in daemon mode. I've got nfs mounts on a system in one datacenter and I've more nfs mounts on a different system in a second data center. I'm trying to sync of of the nfs mounts and I'm getting the errors seen below. I've got five other nfs mounts running exactly the same way with out an issue.

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, 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:

Hi, Yiping! I am not sure of the answer to your question, but I am forwarding it to the LLVMdev list where I am sure someone will be able to answer you. Please send development questions directly to LLVMdev and you will get a response quicker, as it is read by many LLVM developers. ----- Forwarded message from Yiping Fan <fanyp at cs.ucla.edu> ----- Date: Mon, 31 Oct 2005 17:20:24 -0800

Hey all, i do have the following problem: i set up a PDC with Samba with an LDAP backend. Everything works fine but the machine account lookup. If i try to logon to the domain i have to create the machine account in ou=People,dc=testing,dc=de. Everything works fine with this. But if i create the machine account in ou=Computers,dc=testing,dc=de and change all suffixes according to this the