similar to: RE: SIP i.e. Is something broken?

I've searched the site with google, but can't think of the magic words I guess. I got a "swap out" TDM30 today to replace my buzzy one. I swapped it with the older one, swapped out the FXS modules, hooked it up to the computer's power supply, and booted, but the wcfxs driver won't load--it gives me the standard "Cannot init module" types of errors that

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

Hello, I had a partner of mine present a Centrex 21 brochure and ask how many of those features can I fulfill. There is nothing out of the ordinary, it's stuff like call hold, call forward, 3-way calling, etc. Has anyone assembled a how-to that shows how to configure PBX or Centrex type functionality? I found one in the voip-info wiki but only a couple of topics were filled out. Regards,

I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.

Hello, Has anyone had any good or bad experiences with the new Sipura 2.0 firmware? The 1.0.33 works pretty well but there are a few more features I'd like to have. Regards, Christopher J. Wolff, VP CIO Broadband Laboratories, Inc. http://www.bblabs.com

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

Hi, We try ti hide the caller id at calls trought E1 in EuroISDN (Spain) using restrictcid=yes and doesn?t work. What can I do, thaks Pedro -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: mi?rcoles, 31 de marzo de 2004 12:00 Para: asterisk-users@lists.digium.com

Hi Emil, First, thanks for the response. As you mentioned, a lot of times tryCatch does the work, as you can return a value. However, sometimes it is useful to assign several variables when an error occurs. You could do it with <<-, but I prefer to reduce it's usage unless completely necessary. I guess that the attachment was missed in the moderation. Here it is the function:

Prompted by the recent discussion on the mailing list regarding the Asterisk development and release process (or lack thereof), John Todd, Thorsten Lockert, Brian K. West, and myself have put together a plan to address the most significant two legitimate concerns that have been expressed regarding these processes. Specifically: Concern #1: Asterisk release schedules and path to 1.0.0 Asterisk

Hi Ernest, I''m cc''ing the win32utils-devel mailing list in case anyone has any ideas. --- Ernest Ellingson <erne at powernav.com> wrote: > Daniel Berger wrote: > > >Hi Ernest, > > > >Me again. I just realized that it''s just a bug in > my > >documentation (duh). You can pass arguments to the > >Service.start method.

On 20/04/15 17:45, john wrote: > > > > Is this wheezy ? if so, it might be an idea to use backports, this > will get you 4.1.17 which is still in development, 3.6 is now EOL > OK, I understand a bit better where your problems lie. I would still use backports, supported code is (hopefully) better code :-) > > > I'd be willing to do that if it got me support

Hi Ernest, To start: I don't see an attachment, I think they're not (always) allowed on this mailing-list. If you want to send something, text is your safest bet. But regarding the issue of tryCatch: I think you're not fully using what it already can do. In almost all circumstances I've encountered the following works fine: res <- tryCatch(expr, error = function(cond) { # a

John Todd's Onlamp article mentions an OpenBSD version as of June 2003. Have I been sleeping while reading asterisk-users? Is it a seperate project or is it just making Asterisk portable? Who is working on this and is it in the main CVS yet? Do they have device drivers ported or just the software parts of Asterisk? I think the software parts would be relatively simple but time consuming.

Hi all, I use asterisk cvs-head from last week and was wondering why my macros are not working like I expected. A macro will be aborted if one application returns error (-1), but some applications (like ChanIsAvail() in my case) do return -1. So when I want to use ChanIsAvail() in a macro, it does not work if the channel is not available, because the macro is not executed on n+101. The macro

Arnaud Quette <aquette.dev at gmail.com> va escriure (2011-02-28 14:48): > > 2011/2/14 Ernest Sales <ersaloz at gmail.com> > > > Arjen de Korte <nut+users at de-korte.org> va escriure: > > > This is a known problem in nut-2.4.3 and should be solved in > > > nut-2.6.0. We have also dropped the megatec and > megatec_usb drivers, > > >

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Hi, I was wondering if there's a function in R that is meant to return a string representation of an object. Basically, it's like print() but it doesn't print anything, it only returns a string. I know there's a str() function but it's not quite the same. I mean a function that returns the same string that print() would display. -- Bye, Ernest

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with