similar to: Text entry by DTMF

Steve, Do you know if there are many differences between the Mexican and Brazilian variant ? Kind regards, Miguel Antonio Date: Tue, 05 Oct 2004 22:51:53 +0800 From: Steve Underwood <steveu@coppice.org> Subject: Re: [Asterisk-Users] RES: Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Has anyone found a solution for asterisk and r2 signaling ? Steve Underwood had given some information saying he had a working asterisk working. I need it to work with Argentina R2 signaling Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040915/3289f694/attachment.htm

On 11/5/07, Chris Lattner <clattner at apple.com> wrote: > Hi All, Hi, > > http://llvm.org/docs/tutorial/ Very interesting tutorial, reading it now :) > Anyone have thoughts or feedback? :) There's a typo in http://llvm.org/docs/tutorial/LangImpl5.html "the Phi node expects to have an extry" s/extry/entry/ Best regards, Edwin

If I make a call to asterisk's r2 E1 I get these errors *CLI> Jan 10 16:16:38 WARNING[1089170112]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Rx bits 0x1 [1/ 1/ 0/ 0] Jan 10 16:16:38 WARNING[1089170112]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Detected Jan 10 16:16:38 WARNING[1089170112]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Making a new call with CRN

Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Most Fax machines do work but I have problems with people having Tobit FaxWare and Shamrock CapiFax. http://www.tobit.com/login/mrd.asp?CategoryID=120 http://www.shamrock.de/ I've got black bars over the pages. In Tobit some content is Ok,

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

Hello, I've been lurking around the mailing list and browsing around on Asterisk-related links while I wait for my X100P to come in the mail. So far I haven't found very much information related to what I want to do with Asterisk. I was wondering if someone could point me in the direction of any work that may already have been done on a project similar to the one I'm trying to

I'm running Asterisk 1.8 from Debian. I have some analog phones connected via a TDM400P. I'm testing them with these simple extensions: exten => 600,1,Answer() same => n,Festival(This is an echo test) same => n,Festival(Hang up or press pound when you are done) same => n,Echo() same => n,Festival(Good-bye) same => n,Hangup() exten

can anybody help me to choose the most reliable fax solution for * . after googling the net i found that there are at least two solutions for this, app_rxfax+spandsp and iaxmodem+hylafax. - what's the differences between these two? - which one's better? why? thanks

I received a recommendation to check out Asterisk, as a platform to host a simple DTMF response system, something like: Setup up VoIP endpoint on Linux/FreeBSD system Answer incoming VoIP phone calls User enters 100#, perl script plays back "foo" User enters 101#, perl script plays back "fum" User enters 102#, perl script looks up something in

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Hi all, I have some probleem detecting DTMF send by a GSM phone, I'm using SIP with ulaw. do you know what are the options to improve the detection ? I'm using asterisk 1.05, is the CVS HEAD version had some improvement about DTMF detection? Florian.

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

I subscribed to this list for about two months before I began posting, so I've got a buttload of email to sift through ... I'm doing this BEFORE I flood the list with my inane questions ... But here goes: I read a reply from one guy to another about recording. The message included this context from extensions.conf: [recordings] exten => 500,1,Festival('Please record your

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Hi all, Anyone have an idea which version of spandsp, libunicall, libmfcr2, libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the latest asterisk 1.2? Would that be the ones listed below? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/