Re: [Asterisk-Users] TE406P - MFC/R2

Fernando Lujan wrote: Guys, I'm trying to configure a TE406P with MFC/R2. here goes my zaptel.conf: span=1,0,0,ccs,hdb3,crc4 cas=1-15:1101 dchan=16 cas=17-31:1101 span=2,0,0,ccs,hdb3,crc4 cas=32-46:1101 dchan=47 cas=48-62:1101 That config is completely wrong. Try following the config in th

Re: [Asterisk-Users] PCI-X PRI hardware

Boris Bakchiev wrote: HI, Does anyone know if there is a PCI-X 4 port PRI cards available on the market? If so, have anyone used it and how reliable they were? Any help is appreciated... The 3.3V Digium cards and the Sangoma cards work in PCI-X slots. However, I still haven't seen a PCI-E

Re: [Asterisk-Users] MFC/R2 for Voice and Data

Carlos Chavez wrote: I have a question for people who use R2. I know it is possible to use the same E1/T1 for voice and data when you use a TE110P with ISDN. Is it possible to do the same when the link uses R2 for signaling? I know that you can use Unicall to handle R2 on voice, but h

Re: [Asterisk-Users] Prices of g729 codec

Andrew Kohlsmith wrote: On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote: Again, 10k channels you'll have a half dozen MaxTNT boxes terminating DS3s. Your fixed costs will already be significantly higher and that little $10 license fee is included in that. Its not $10, wh

Re: [Asterisk-Users] Prices of g729 codec

Cory Andrews wrote: Voiceage in Montreal is supposed to be working on an open source G.729A codec, although it mentions only that they allow developers to freely use their G.729(A) codec object code for non-commercial purposes. Lots of good codec related info here http://datacompression.info/Sp

Re: [Asterisk-Users] Prices of g729 codec

Kevin P. Fleming wrote: - trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote: speex isnt in all ATAs and other things. So if its not there it offers worse compression since the call wont go through :P And it's not there because it is significantly more computationally intensive

Re: [Asterisk-Users] Quad T1 Card

Andrew Kohlsmith wrote: On Wednesday 07 June 2006 11:19, Matt Florell wrote: turned on and off as needed by it's firmware. The Digium card uses an Oki chipset that has a smaller echo tail length and is hard-coded into the firmware so you cannot change it. Actually Oki's just the fab.

Re: [Asterisk-Users] Quad T1 Card

Kevin P. Fleming wrote: - Mike Fedyk <[EMAIL PROTECTED]> wrote: Will it have a 1024 tap echo can on all 96 channels? What about 8 T1 support like sangoma? Those are completely unrelated questions; there is no need for an 8-span echo can module when there is no 8-span T1 card :-

Re: [Asterisk-Users] Quad T1 Card

Michael Collins wrote: Kevin P. Fleming wrote: According the Sangoma data sheet, the Octasic part _is_ the DSP (which it is, in a logical sense). The board does not relieve Asterisk/Zaptel of any additional burden beyond echo cancellation and tone detection at this time;

Re: [Asterisk-Users] Fun with Echo

Rich Adamson wrote: Eric "ManxPower" Wieling wrote: The number of taps the EC has to deal with is the delay on the PSTN side. I can't imagine echo is more than a few ms in modern TDM networks. This latency has NOTHING to do with VoIP latency, since the echo must be canceled BEFORE it gets

Re: [Asterisk-Users] Unicall acting really funny

Hi Joao, One of the *big* advantages of the Tormenta 2 cards over the more recent cards from Digium is the driver actually counts various errors, and lets you see link problems easily. The FAS error count is increasing, which means you have a link problem. There, that was easy. With a TE411

Re: [Asterisk-Users] Fun with Echo

tions. Actually that's untrue. I think (hope) that Steve Underwood will jump in here and correct me, but it's my understanding that the only real reason why the software echo cancellers available in Zaptel don't work as well as the hardware echo cancellers from Tellabs and the

Re: [Asterisk-Users] grandstream GXV-3000

Mike Fedyk wrote: Or any polycom phone that has speakerphone like the IP501 and IP430. Time Bandit wrote: Can you, or anyone else comment on the speakerphone ability of the GVX-3000 ? We run the GXP-2000's and for the most part are happy with them, but for upper management we're looking at

Re: [Asterisk-Users] Hard drive write cache

shadowym wrote: The cold hard truth is that if Asterisk cannot achieve 99.999% uptime without becoming much more expensive that a traditional PBX then it is not a viable alternative. Even elcheapo Key systems are rated for five nines. Even massive redundant public exchanges struggle for 99.9

Re: [Asterisk-Users] GXP-2000 Audio Quality

Welcome to the wonderful world of VoIP, where people are eager to move from 8kbps G.729 to 6.3kbps G.723.1, and accept a substantial drop in voice quality, and then throw over 20kbps of RTP, IP and related overhead on top of them. Isn't IP wonderful? :-) Regards, Steve Daniel Salama wrote: W

Re: [Asterisk-Users] FAX + Digium + SpanDSP

Pibix wrote: Hi, Anyone using SpanDSP with Digium TDM o TE cards to receive and email Faxes? No. Nobody ever uses this stuff. I just write it to waste my spare time. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asteris

Re: [Asterisk-Users] T1 Copper or T1 Fiber Line

Chad Osmond wrote: T1 PRI's are (almost?) always copper. Pretty much always copper by the time you see them. They might be fibre from the box on the wall to the telco's luxurious mansion, but what you connect to is almost certainly two twisted pairs, or two thin co-ax cables (mostly the tw

Re: [Asterisk-Users] MOS Scores and LCR

trixter aka Bret McDanel wrote: On Sat, 2006-06-17 at 10:16 +0200, Florian Overkamp wrote: There are ways to guesstimate MOS scores on a call by continuously getting some decent statistics from the jitterbuffer. We've had an intern do some work on this using IAXclient. http://www.speakup.

Re: [Asterisk-Users] MOS Scores and LCR

trixter aka Bret McDanel wrote: On Sat, 2006-06-17 at 23:25 +0800, Steve Underwood wrote: Calling MOS totally subjective is rather strange. Telephony only has to meet subjective goals. In reality, MOS is pretty objective, as it is a carefully controlled experiment across enough subjective

Re: [Asterisk-Users] Where's the Fiber

[EMAIL PROTECTED] wrote: Where's the Fiber? I was reading about T1 lines and came across this statement.. It basically said T1's are made up of copper...Wasn't T1 made up of Fiber? Is the new trend to move T1 away from fiber and use copper? Commerical T1 systems were introduced in about 1960

Re: [Asterisk-Users] sangoma unicall m2rfc

Anton Krall wrote: Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and

Re: [Asterisk-Users] ECHO Tutorial

Seth Remington wrote: On Mon, 2006-06-19 at 18:45 -0400, Gary Reuter wrote: On 6/19/06, Daniel Salama <[EMAIL PROTECTED]> wrote: Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? This paper by Cisco is a g

Re: [Asterisk-Users] Unicall acting really funny

Hi Joao, What is the T3 timeout set to? If it is short, try making it something like 15000 (it is in milliseconds). Your T3 timeout problem could be due to slow reponses from some destinations. For your second issue, the following is definitely a protocol error: UniCall/1 <- 1 off [2

Re: [asterisk-users] Clocking Multiple T1 Cards

Andrew Kohlsmith wrote: On Monday 24 July 2006 12:11, Shaw Terwilliger wrote: Thank you; this is the kind of information I was looking for. The wiki and other documents told me exactly what the configuration options did, but I didn't know what kind of timing configuration was right for mult

Re: [asterisk-users] Rxfax and squashed TIFF

Garth van Sittert wrote: Hi All Just wondering if anyone knows of a solution to the squashed tiff problem with spandsp (or rather Windows Image Viewer) other than converting to a PDF. I find the PDF image quality is not nearly as good as the original TIFF. Apparently the Windows Image View

Re: [Asterisk-Users] Which ATA to test T.38 ? What about Linksys 3102

Thomas Kenyon wrote: Peder @ NetworkOblivion wrote: When I looked several months ago, the only Sipura that supported T.38 was the SPA-2100. I haven't searched in a while, but I think it is still true. We go directly from a Cisco gateway to the SPA-2100 and it works great. It is the only A

Re: [asterisk-users] CSTA support for asterisk

Joshua Colp wrote: Hi, Can anybody tell me that is their CSTA support for asterisk Due to the fact that nobody seems to know what it is - I'd say no. Can you shed any light on what it is? In this context "nobody" would seem to mean you. Surely most people who have done anything

Re: [asterisk-users] Unicall stack, right versions?

Barzilai wrote: Last night I started compiling all the components of the Unicall stack. So far I've been able to successfully do a "testcall". A couple of questions: 1) If you download the "snapshot" libraries, a funcion that used to be called "dtmf_put" now has been changed to "dtmf_tx_put",

Re: [asterisk-users] Unicall stack, right versions?

Barzilai Spinak wrote: Thank you Steve. About the configs in Asterisk... I confess that I'm new to the code so I still need to read more. I didn't know about ast_config() About the hardcodedness of the countries... that seems to be the "problem". Everything is too oriented to "my country wor

Re: [Asterisk-Users] ADPCM - vs - G.726

Whisker, Peter wrote: I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is

Re: [Asterisk-Users] If a packet is dropped, is it possible to replay the previous packet in Asterisk?

Gabriel Afana wrote: At the VON expo, my SIP workshop mentioned this as a technique to help cover up dropped packets. Has anybody heard anything about Asterisk being able to do this? - Gabe Simply repeating packets is a terrible technique. Asterisk does something much better. Look for "PL

Re: [Asterisk-Users] Alarm on Unicall

Hi, You will notice the reports are of alarms being cleared. If this just happens at start up, it may just be reports of the E1 initially setting down after the link is established. If you keep getting alarm reports during normal operation, your link is probably unreliable. Steve acriollo

Re: [Asterisk-Users] Span monitoring

Kevin P. Fleming wrote: Wai Wu wrote: Does Asterisk have builtin (T1 or E1) span monitoring? If a span goes down, will asterisk know about it. Personally, I would like to have a event generated through the Manager API interface. Have you actually tried this? It takes all of about 10

Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?

Don Pobanz wrote: Adolfo R. Brandes wrote: Lee Howard wrote: However, based on the comments you give I'd suspect that you're having what people seem to be calling "frame slipping". There seem to be some motherboards that react poorly with Zap cards (or their respective drivers) and cause

Re: [Asterisk-Users] R2 protocol error

Hi Dennis, Update to libmfcr2-0.0.3 pre9. I made a slip in pre8. Sorry. Steve Dennis Nacino wrote: Hi, I have three R2 installation on different carriers, all shows the same inconsistency at varying degree. But, on most test calls we made, it reaches T3. The worst part of these, the carri

Re: [Asterisk-Users] Compatible Asterisk Connectivity Cards : Sangoma

Pete Barnwell wrote: On Mon, 2006-04-03 at 07:19 -0500, Brian Roy wrote: On 4/2/06, Heidi Mendoza <[EMAIL PROTECTED]> wrote: Hello List! I wanted to share to everyone the following compatible connectivity products that my company installed in our Asterisk

Re: [Asterisk-Users] WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)

Ronald Wiplinger wrote: Matt wrote: Hey all, I recently installed the Sphinx voice recognition software, and let me just say that it rocks... major time. Now... a question for those who may be more experienced then I.. Is there anyway I can have Sphinx be listening WHILE I am playing a pro

Re: [Asterisk-Users] Digium cards, so disappointing !

Kevin P. Fleming wrote: Rusty Dekema wrote: If this works, I don't see why a fax transmission wouldn't work. Is it because the fax protocol doesn't have error correction? Is that even true? FAX transmission is massively more complex than modem transmission. At higher speeds, it invol

Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, so disappointing !)

Kevin P. Fleming wrote: Jeff Gustafson wrote: Is there any reason an easier implementation of the same, basic, idea could be created for the "Asterisk generation?" According to a quick search of H.100 it's "just" a TDM bus. It handles 2,048 full duplex calls. Would a lightweight

Re: [Asterisk-Users] Digium cards, so disappointing !

Begumisa Gerald M wrote: Hi Steve, Thank you for your very enlightening message! On Sat, 15 Apr 2006, Steve Underwood wrote: [...] > modem it must be applied end to end by the modems themselves. The > real killer, though, is imperfect timing. [...] > and its n

Re: [Asterisk-Users] Digium cards, so disappointing !

Remco Barende wrote: On Sat, 15 Apr 2006, Kevin P. Fleming wrote: Actually, I did. During a FAX transmission, there are many shifts to different carriers and signaling rates as pages are transmitted and acknowledged. It is _not_ as simple as a single carrier, like a normal data modem connecti

Re: [Asterisk-Users] Digium cards, so disappointing !

Remco Barende wrote: record the sound fax machines make when negotiating (specifically the part where they try to negotiate anything above 9600 baud) and make a provision in asterisk (an extra letter added to the Dial command?) that will make Asterisk monitor the channel and listen for the fax

Re: [Asterisk-Users] Digium cards, so disappointing !

Remco Barende wrote: Hmm not so sure of that. I have an HP all-in-one thingy. It is not possible to set the TX/RX speed hard in the config at a certain speed. Through the developers menu in the beast it is possible to do this temporary. Faxing at max 9600 bps works, anything higher fails mis

Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,

Doug Lytle wrote: Nicholas Kathmann wrote: Both hylafax and * are on the same machine and using the same PSTN interfaces (whether T1 or TDM). It uses iaxmodem to communicate between the two systems (imagine a softmo

Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards,

|Subject: Re: Faxing and PCI (was Re: [Asterisk-Users] Digium cards, | |Steve Underwood wrote: |> Doug Lytle wrote: |> |>> Nicholas Kathmann wrote: |> |> |> If you need to tweak gains something is seriously wrong. |> |The 2 fax machines that I was having problem with were faili

Re: [asterisk-users] Faxing with asterisk

Fabio Mosti wrote: > Hi All, > > I need to setup asterisk to receive fax. > > I'm try Spandsp (opensource) and Attrafax (commercial) both on > asterisk 1.4.23) but the results are disappointing. > with spandsp many times the fax arrives cut. > with Attrafax i have some problem. > > Anyone have any

Re: [asterisk-users] Faxing with asterisk

Steve Underwood wrote: > Fabio Mosti wrote: > >> Hi All, >> >> I need to setup asterisk to receive fax. >> >> I'm try Spandsp (opensource) and Attrafax (commercial) both on >> asterisk 1.4.23) but the results are disappointing. >> with spand

Re: [asterisk-users] SpanDSP question for Steve

Michael wrote: > Hello- > > Firstly thanks very much for the work you have put into SpanDSP and the time > you spend to assist people here :-) > > I am currently running SpanDSP 0.0.5 with Call Weaver. Is there any or > sufficient gain to be had from upgrading SpanDSP? > I don't follow the sta

Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

Hi, Asterisk Asterisk wrote: > We've had a 65% success rate across the board (actually 35% > incorrect). I'm working on bringing that up to 85% or better. Good gender recognisers get >90% success on PSTN lines. In restricted contexts they can get up to 98%. These figures are not entirely honest,

Re: [asterisk-users] dahdi wcb4xxp and fax

Lee Howard wrote: > stoffell wrote: > >> I wanted to switch from my current setup (mISDN) to the native dahdi >> with b410p support (wcb4xp). All works fine for normal phone calls but >> not for faxing. Faxes are distorted, if arriving at all, and hylafax >> logs the usual bad stuff (HDLC fra

Re: [asterisk-users] Silk for Free

Dean Collins wrote: > > http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio_codec.html?tk=rss_news > > any thoughts? > They have said it will be royalty free, but they have said little else. From discussions with Skype people in the last few days they seem very reluctant to

Re: [asterisk-users] Wideband g711-HD vs. g711.1?

Philipp von Klitzing wrote: > Hi there, > > has anyone seen specifications of the codec g711-HD? This is right now > spreading fast in the wake up CATiq (the DECT successor), for example in > the AVM products (www.avm.de). > > Is this a re-branded g711.1 (rfc5391) and therefore compatiable with i

Re: [asterisk-users] Wideband g711-HD vs. g711.1?

Kevin P. Fleming wrote: > Steve Underwood wrote: > > >> CAT-iq supports G.722 for wideband voice. the CAT-iq web site describes >> this as "CD quality". I guess the person who wrote that has severely >> impaired hearing. :-) >> > > Maybe t

Re: [asterisk-users] faxing via linksys SPA3102 half page goes through

Marco Signorini wrote: > Joseph wrote: > >> On 03/04/09 15:44, Marco Signorini wrote: >> >> >>> Hi Joseph. >>> I've spent some time tuning the SPA3102 FXS line input and output gain >>> and I think that this is an important variable. >>> Let's try to record incoming and outgoing fax tone

Re: [asterisk-users] Silk for Free

Wilton Helm wrote: > >12kHz isn't really enough for high quality voice, and the extra bit > >rate needed to push the bandwidth to 15kHz is small. Also, a deep man's > >voice looses something when you cut off at 70Hz. > > I'm not sure that this isn't stretching things a bit. There are no > hands

Re: [asterisk-users] Silk for Free

Kevin P. Fleming wrote: > Steve Underwood wrote: > > >> Good engineering of standards is about building them for the future. >> Cutting off the bass at 70Hz is far less of a limitation than cutting >> off the high end at 11kHz, but why do it in the codec? Why not le

Re: [asterisk-users] Silk for Free

Steve Underwood wrote: > Kevin P. Fleming wrote: > >> Steve Underwood wrote: >> >> >> >>> Good engineering of standards is about building them for the future. >>> Cutting off the bass at 70Hz is far less of a limitation than cutting >

Re: [asterisk-users] Faxing success rate on PRI

Marco Signorini wrote: > Hi Gordon, thank you for your answer. > > It's not mandatory to use an external box to handle the PRI. I was > thinking to use a Patton device instead of a TE120P just because I would > like to be able to switch to T38 in the near future or if working with > inband faxes wi

Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6

Santiago Gimeno wrote: > I finally solved the issue by changing the resolution and the width of > the TIFF file to one that is accepted by the fax standard. In my case > I changed to a resolution of 96x96 and a width of 1728. > > Now I am able to send faxes, but something weird is happening, the

Re: [asterisk-users] Ast/Hyla/IAX Scalability?

David Backeberg wrote: > On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson > wrote: > >> On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg >> wrote: >> >>> Again, you'll find people arguing that their voip solution has as low >>> of a failure rate as a hardware solution. I'm jealous. M

Re: [asterisk-users] Asterisk 1.6 ReceiveFAX problem

MaxGao wrote: hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called w

Re: [asterisk-users] t38 iax trunk

dubravko caric wrote: > Hi all, > > I have a question regarding using T38 for fax sending and here is my > scenario: > > fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk > #2 -> SIP ATA (T38 enabled) -> fax > > My question is, how can I know if I'm really using T38? is T38 >

Re: [asterisk-users] Ast/Hyla/IAX Scalability?

VB wrote: > If you using cisco why don't you use fax on/off ramp it works quite well. > Then you can do with the fax file whatever you want. > > >From other point of view I did connected 1.6.0.6 with spandsp-0.0.5 to PRI > and receivefax seems to be working ok. The connect speed is low somewhere >

Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?

Vlasis Hatzistavrou (KTI) wrote: > Olivier wrote: > >> Hi, >> >> I've been playing with T.38. >> >> I observed that mostly but not always, it's the "calling endpoint" that >> reINVITE the other party to drop current SIP/G711 session and start a >> new T.38. >> But sometimes, it's also the call

Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?

party cannot initiate it, too. > >> > >> Best regards, > >> Vlasis Hatzistavrou. > >> > Steve Underwood wrote: > > Hey, why bother looking at a spec when its so much more fun to > make it > > up as we go along? > > >

Re: [asterisk-users] T.38 - Which endpoint shall reINVITE ? caller or callee ?

Hi Olivier, Olivier wrote: > > T.38 says that if the call starts in audio mode it is the called end > which should initiate a re-invite to change from audio to T.38. This > makes sense, as that is the end which has the best chance of figuring > out if a FAX machine answers the call

Re: [asterisk-users] Wideband g711-HD vs. g711.1?

Philipp von Klitzing wrote: > Hi! > > >>> has anyone seen specifications of the codec g711-HD? This is right now >>> spreading fast in the wake up CATiq (the DECT successor), for example in >>> the AVM products (www.avm.de). >>> >>> >> Googling for G.711-HD only produces hits about AVM

Re: [asterisk-users] Ast/Hyla/IAX Scalability?

David Backeberg wrote: > On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies wrote: > >> While we have your attention Steve (Underwood) do you have a >> high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We >> currently use 0.0.4 with a very high su

Re: [asterisk-users] OpenBTS chat with David A. Burgess

Tzafrir Cohen wrote: > On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: > >> Hi, >> >> "The OpenBTS Project is an effort to construct an open-source Unix >> application that uses the Universal Software Radio Peripheral (USRP) >> to present a GSM air interface ("Um") to standard GSM hands

Re: [asterisk-users] codec payload size

ContactTel Business wrote: > People should use .020 ms sample rates for RTP as it's the standard. 0.030 > was i think the old SPA implementations which caused MR, Roboto kind of > grabling. > > > You should find a way to patch your sip core i assume, but dev's could tell > you where. > > We offer 0

Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?

Tim Nelson wrote: > - "Wilton Helm" wrote: > > > >If half-duplex audio is good enough for you, sure. > > You've lost me there. I am not aware of a modem that is for sale > today that is half duplex. (OK some support a couple of minor half > duplex modes). All state of the art modem proto

Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?

Martin wrote: > I wonder why people don't get it ? X100P is a winmodem was and always will be. > What makes you think anyone doesn't understand that? The problem is the chip on the X100P isn't made any more, and X100P cards are no longer so plentiful. You'll notice the price is going up. They

Re: [asterisk-users] [Zaptel] Why no driver for PCI voice modems?

Tzafrir Cohen wrote: > On Thu, Apr 02, 2009 at 08:28:44PM +0800, Steve Underwood wrote: > > >> Several Winmodem chips are still readily available, and so are cards >> containing them. What is missing is someone putting the effort into >> making drivers for them. &g

Re: [asterisk-users] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA

Hi, There are at least 2 R2 protocol variants in Colombia - one used by land lines, and one used by the cellular networks. Unicall implements both, and I think both have been used successfully by people in Colombia (I seem to remember debugging with people there long ago). Which is the protoco

Re: [asterisk-users] Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA

r2 eventually times out > waiting for the category. > > Giovanny, if the problem persist after my recommendation contact me > off-list to arrange a debug session. > > Moy > > On Thu, Apr 9, 2009 at 8:02 AM, Steve Underwood wrote: > >> Hi, >> >> There are

Re: [asterisk-users] T.38 ATAs

Ian wrote: > Hello > > I am going to try the new Digium Fax for Asterisk product. I'm planning > to connect fax machines to Asterisk (currently 1.6.0.9) via T.38 ATAs. > I'm looking at Grandstream HT502 or Linksys SPA2102 ATAs. If anyone has > any experience with these devices, or other recommen

[asterisk-users] FAX reliability

Hi, Most people using iaxmodem + HylaFAX or spandsp from an Asterisk application get satisfactory results for both transmitting and receiving FAXes. However, quite a few people find FAX receive is reliable, but FAX transmit can be flaky. Its box dependent. The cause of this has been known sinc

Re: [asterisk-users] FAX reliability

Hi Lee, Lee Howard wrote: > Hi Steve, > > Steve Underwood wrote: >> In chan_dahdi.c there is now code that extends the buffering inside >> dadhi when a FAX is detected, and puts the buffering back to normal >> at the end. This isn't really a cure - its more of

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

Lee Howard wrote: > David Backeberg wrote: > >> It may be possible to use hylafax, but >> I don't know how or why you would. >> > > The reason *why* is generally due to support issues. > > For one, HylaFAX probably has a better T.30 implementation in its Class > 1 driver than does app_fax.

Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1

David Backeberg wrote: > What I was specifically getting at in the context of that response was > a comparison of dynamic modem pool versus fixed-size modem pool. When > faced with the choice between a fixed-size modem pool or one that > would grow or shrink dynamically with demand, I think the dyn

Re: [asterisk-users] Digium Fax for Asterisk questions

Michael wrote: > 1. What is the difference between Asterisk v1.6 RXfax/TXfax and Digium Fax > for > Asterisk? > > 2. Is Free Fax for Asterisk identical to Fax for Asterisk nonwithstanding the > 1 channel limitation? > > 3. Can any purchase of Fax for Asterisk count as channel 2+, when used in >

Re: [asterisk-users] Polycom wideband codecs?

mgra...@mstvp.com wrote: > 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? > Which Polycom supports G.722.2? I think they are only supporting G.722,

Re: [asterisk-users] Looking for good IAX ATA

Jeff LaCoursiere wrote: > On Sat, 25 Apr 2009, Yahya Mohammad wrote: > > >> On Wed, Apr 22, 2009 at 09:20:05PM +, Jeff LaCoursiere wrote: >> >>> I have been wondering - if you ran your SIP traffic over VPN tunnels, what >>> would the state think of that? They obviously won't be able to

Re: [asterisk-users] Digium fax failing

Michael wrote: > On Mon, 27 Apr 2009 03:40:12 you wrote: > >> Slightly off topic, but M$ is worth billions because they started in 1976 >> or so, became the de facto standard, and were pretty cutthroat in the way >> they do business. They have a profit motive and have always taken the path >> t

[asterisk-users] POS modems

Hi, If anyone is interested in the low speed modems needed for POS applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I had some spare time while travelling, and finally got the V.22bis code I started a long time ago into a start where its basically functional. I'm now looki

Re: [asterisk-users] POS modems

Thomas Kenyon wrote: > Steve Underwood wrote: > >> Hi, >> >> If anyone is interested in the low speed modems needed for POS >> applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I >> had some spare time while travelling, and finally got t

Re: [asterisk-users] Problems receiving some faxes in T.38

Hi Santiago, Santiago Gimeno wrote: > Hello, > > We have been working with the ReceiveFax application for some weeks > now in order to receive faxes in T.38 and it works fairly well, but > there are some faxes that for some reason we are not able to receive > correctly. > > The asterisk version

Re: [asterisk-users] Problems receiving some faxes in T.38

Santiago Gimeno wrote: > Hi Steve, > > Thanks for the answers. > > Comments inline. > > 2009/5/20 Steve Underwood : > > >> Did you draw that arrow in the wrong direction? The side answering the >> call should send the first V.21 signal. >>

Re: [asterisk-users] Maximum cable length for analog phone from FXS port

Tzafrir Cohen wrote: > On Tue, May 26, 2009 at 05:39:46PM +0200, randulo wrote: > >> On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas wrote: >> >>> I run my analog telco over cat5, but that's in-house and definitely not >>> 3km. That sounds really far for current loop stuff. >>> >>

Re: [asterisk-users] OPENR2 in Thailand

Hi Peter, Thailand is similar to China, except for two things. - Some places require that billing pulses be generated. - There may be places using DTMF instead of MFC. The first issue is definitely the case. The second might just be the false reporting of issues. People definitely used Unicall

Re: [asterisk-users] fax / t38 gateway

Olivier wrote: > Linksys PAP2 or 3102 for instance > or Patton M-ATA > > In fact, I would say most analog gateways with FXS port should also > support T.38. > In this case, your setup would be : That list rather poorly supports your argument. The PAP2 and the PAP2T do *not* support T.38, despit

Re: [asterisk-users] fax / t38 gateway

Olivier wrote: > > > 2008/10/24 Wilton Helm <[EMAIL PROTECTED] > > > > I've been following this thread and trying to sort out what is > wanted, what is available, and why. Comments to the following > would be appreciated and might be useful to others. >

Re: [asterisk-users] fax / t38 gateway

Benny Amorsen wrote: > Steve Underwood <[EMAIL PROTECTED]> writes: > > >> That list rather poorly supports your argument. The PAP2 and the PAP2T >> do *not* support T.38, despite numerous arguments you'll find to the >> contrary. Personally I beli

Re: [asterisk-users] fax / t38 gateway

Benny Amorsen wrote: > Steve Underwood <[EMAIL PROTECTED]> writes: > > >> Even the big floor standing office MFPs typically only offer T.37 or >> T.38 only through an expensive option card. >> > > Medium MFP's almost all support T.37. They call

Re: [asterisk-users] fax / t38 gateway

Hi, A lot of people talk about grooming to make VoIP work smoothly, not just for FAX. However, most people can only achieve grooming in one direction. Their ISP will not cooperate, and groom what is sent to the subscriber. Unless you just keep your DSL link very lightly loaded, by doing no bro

Re: [asterisk-users] fax / t38 gateway

JD wrote: > Gordon Henderson wrote: > >> but it's very do-able, given good Internet connections. >> > [...] > >> I think your statements were just a bit too strong - I agree >> wholeheartedly about the V. protocols and copper, but I've found in >> practice that faxing over IP is not ju

Re: [asterisk-users] fax / t38 gateway

Jonn R Taylor wrote: > I have been able to repeat the results at other locations. The location that > has 26 pages is a linksys PAP2T our accounting person uses remotely to fax > stuff to the office. The ATA is behind a DIL-625 router with QOS on a DSL > line. > > I can send faxes from my test

Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

Wilton Helm wrote: > I'm a bit puzzled, also, having implemented ulaw and alaw in an > embedded application. Each can be done with a 16 Kbyte table in about > 0 time with no errors. There are probably tricks that will cut the > table down by 2 or 4 X for a small cost in CPU cycles. The invers

Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

Matthew Fredrickson wrote: > Actually, with the way caching is done on nearly all modern processors, > it is debatable whether or not a look up table is the optimal way to do > the conversion, at least on such a simple codec such as ulaw or alaw. > In fact, the amount of time it takes to fetch m

Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)

Benny Amorsen wrote: > Steve Underwood <[EMAIL PROTECTED]> writes: > > >> In spandsp I do the G.711 conversions algorithmically. Most modern >> processors have a "where is the top 1" instruction, and that reduces the >> calculations to something ve

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