similar to: Algorithmic delay

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

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we

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:

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,

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

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

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

Hi, I was hoping to get some clarification on the meaning of algorithmic latency in Speex. For wideband signals, algorithmic delay is specified to be 34 ms (20 ms is the frame size and 14 ms is the lookahead). Does this number (34 ms) account for the double-buffering on the input? Assuming that the computation time and sound wave propagation time are negligible, should I expect compressed

On 01/20/2011 04:26 AM, Steve Underwood wrote: > On 01/19/2011 06:44 PM, LiMaoquan2000 wrote: >> Hi all, >> >> We have discussed so many about sampling rate asynchronous (or offset) >> between rendering (D/A converter) and capturing (A/D converter) of >> most PC soundcards. It seems all acoustic echo cancellers, include AEC >> in speex, can not deal with this

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

Hi all, I would like to announce the availability of an initial test version of a totally software FAX facility, suitable for use with Asterisk. This is a first public test release, so don't expect a solid polished product just yet. People have shown interest in what I am doing, and here is the evidence that it is not vapourware. If the notion of a software FAX machine is new to you,

The last thing that I read about it was: Steve Underwood [steveu@coppice.org] wrote on Sep 3: >> Is E&M designed to work with the E1 driver code? I think probably not. I >> had to fix some things to get proper access to the CAS signaling bits >> when I implemented MFC/R2... So, apparently he implemented it. I was trying to contact Steve, but he isn't answering me. Does

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/

Hi, Does anyone had the problem of asterisk SendFax + spandsp sending only the first page of a multi-page TIFF file? Seams to be related to spandsp ECM config. Any thoughts about it? Thanks, Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121005/ac471600/attachment.htm>

What is the unsubscribed address? Thanks Michael -----Original Message----- From: Steve Underwood [mailto:steveu@coppice.org] Sent: Friday, February 18, 2005 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Re: quadbri and spandsp You need to use the caller parameter. Something like: Channel:Zap/G1/XXXXXXXX Application:txfax

This message is pointed directly to Steve Underwood. I tought it would not be nice to directly email him with a question that interests a good part of the Asterisk community, so here it is. :) Steve, remember a few days ago when we discussed about issues on Asterisk 1.6.1.13 and T.38 fax reception? Well I opened an issue on Mantis (https://issues.asterisk.org/view.php?id=16756) and turns out it

Hi, I've kind of ported a DTMF text extry method I wrote some time ago for Dialogic. It is now a semi-working Asterisk app. I've still got to clean up some stuff in how Festival is used to read back what is entered, and then I think it should be OK. Would anyone here find this useful? It takes an entered string, and returns it as a variable. It might sound clunky, but for short entry

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?