similar to: Re: FAX, IAX and *....Maybe I'm dreaming...:-) (Carl Youngblood)

Hi, It would be great if the IAX protocol will be able to tranfer fax data (even converted in another format) between Asterisk boxes, using low bandwidth codecs like GSM. I know that this is possible only with the G.711 now (passing faxes using the audio stream), but.... maybe in the future...some native support will permit this. Just a thought, Dan

Greetings to all, We are new to the Asterisk community and really appreciate the willingness you have to share knowledge and help one another. We are building one of the Nations first fiber to the home networks here in Provo and have selected Asterisk to test as our phone switch. Newbie kinds of questions: Is there any web-based tool that would allow non-technical personnel to add new

John Todd wrote.... > Cisco has an 802.11 phone called the 7920, which is apparently > shipping now. It is very expensive (>$550 USD) and only runs SCCP at > the moment, which is Cisco's proprietary VoIP protocol. However, if > it falls in line with some of Cisco's other high-end VoIP equipment, > that means it should have a trailing-edge SIP image running by

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

We are working on a new implementation of asterisk. We are using a fiber-served WorldWide Packet switch at the home that incorporates a VOIP T2 switch that feeds 2 POTS connections. We are told that the T2 is programmed with code that follows the PacketCable spec. This version has a problem with the 'congestion' message which is based on the MGCP RFC (2705) spec. This causes the T2 to

On Thu, Jul 14, 2016 at 11:39:37AM +0200, Alexandre wrote: > Hi, > > I had to review a samba setup recently where people experienced strange > things. > Basically they more from a solaris on phisical machines environment > (locally hosted) running an old version of samba (2.x or even 1.x) to a Red > Hat virtualized environement runing 3.6 (remotely hosted). > The link

CarlWoodward (cjwoodward at gmail.com) has invited you as a friend on Quechup... ...the social networking platform sweeping the globe Go to: http://quechup.com/join.php/aT0wMDAwMDAwMDA5NTI5MTMxJmM9OTkyMDg%3D to accept Carl''s invite You can use Quechup to meet new people, catch up with old friends, maintain a blog, share videos & photos, chat with other members, play games, and

Greetings. I have a scenario I would like some advice on. I have a 100,000 square foot building that we will be moving some work crews into. It has offices on each end of the building and a fiber line between them. I currently have an asterisk 1.2 system in place and about 30 phones. My problem is they want a few fax machines out in the warehouse area where I currently have no wiring for

Hi, I had to review a samba setup recently where people experienced strange things. Basically they more from a solaris on phisical machines environment (locally hosted) running an old version of samba (2.x or even 1.x) to a Red Hat virtualized environement runing 3.6 (remotely hosted). The link between clients and server is really good (2x100Mb/s fiber) and browsing shares and opening office

On Thursday 11 Jun 2015, Luca Bertoncello wrote: > Now my problem is to check in my dialplan if the peer, that originate > the call, is reachable, and if not, to give an error... > > Is there any function to know if the peer is reachable? The peer that *originated* the call *must* be reachable, by definition ..... -- AJS Note: Originating address only accepts e-mail from list!

On 06/11/15 09:56, Luca Bertoncello wrote: > Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > >> On Thursday 11 Jun 2015, Luca Bertoncello wrote: >>> Now my problem is to check in my dialplan if the peer, that originate >>> the call, is reachable, and if not, to give an error... >>> >>> Is there any function to know if the peer is

Hi again! About my previous E-Mail... I though about it and I think, that maybe I'm just very stupid... Since I called an INTERNAL number, Asterisk tried to call it. I tried right now to call an EXTERNAL number (using my context [myproxy]) and the behavior is NOT the same... Not 100% correct, but it tries the right way... Now my problem is to check in my dialplan if the peer, that

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Thursday 11 Jun 2015, Luca Bertoncello wrote: >> Now my problem is to check in my dialplan if the peer, that originate >> the call, is reachable, and if not, to give an error... >> >> Is there any function to know if the peer is reachable? > > The peer that *originated* the call *must* be

Hi all, my first post. After years with an old server, a couple of months ago I've installed a new one. We use IMAP. After research, I chosed dovecot. It has been running for several weeks but from one week ago or so, it suddently dies about once per day. The only change done in the server one week ago has been to install ntpdate, running once per day in cron-daily. In the cron

Hi there, I'm in the following directory: ~/Documents/Rstuff/diceFiles/dice_1.1 The directory "dice" is in this directory, with all the usual build files (DESCRIPTION, NAMESPACE, etc). I'm trying to run the following command: R CMD check dice (where "dice" is the name of the package I'm checking), and I get the following: * checking package dependencies ...

Hi everybody, the following alway gives me the error "Fehler in if (File$X.Frame.Number[a] + 1 == File$X.Frame.Number[a + 1]) (File$FishNr[a] <- File$FishNr[a - : Fehlender Wert, wo TRUE/FALSE n?tig ist". Maybe its stupid, but i'm not getting why... Maybe someone can help me. Thanks a lot! for (i in unique(BigFile$TrackAll)) { File <-

Zitat von Guido Falsi <mad at madpilot.net>: > So, trying to bind authentication to originate calls to registrations is > conceptually wrong in the SIP world. Maybe you can do that but that's > not the way the protocols have been engineered to work. Hi Guido, thanks for your answer. Well, I decided to do that, since I have my Asterisk reachable from Internet just for my

hi all, i've this problem with app_txfax. Here's the log of the error: Sep 8 13:28:55 VERBOSE[10750]: -- Attempting call on Zap/g1/2430 for application txfax(/var/tmp/ast_fax-1126178934.10240.1804289383.0|caller) (Retry 1) Sep 8 13:28:55 DEBUG[10750]: Using channel 3 Sep 8 13:28:55 WARNING[10750]: Unable to allocate channel structure Sep 8 13:28:55 NOTICE[10750]: Unable to

On Mon, Nov 24, 2014 at 12:12 PM, John R. Dennison <jrd at gerdesas.com> wrote: > On Mon, Nov 24, 2014 at 12:04:30PM -0600, Les Mikesell wrote: >> >> Umm, yeah. Encrypted protocols would never be compromised.... > > Which do you think is more likely? Someone sniffing a cleartext > credential set on the wire or someone subverting an alleged "secure" >

Its another one of my "If I only had time...damn this sleep thing" ideas, but I really wonder how hard/cost effective it would be to build an open source IP phone or phone adapter (ala ATA). In about 20 minutes of mulling and research, I figure you could do it for about $40 in parts plus coding time... 1 DS80C400 Ethernet enabled microcontroller with built in IPV4/V6 stack $10 1