Re: [asterisk-users] Codec G.129 A vs A/B

On 06/04/2010 02:27 AM, Kyle Kienapfel wrote: > http://en.wikipedia.org/wiki/G.729 > Looks like theres A and B and no "A/B" so theres nothing to worry about > What's the point of quoting a page, if you are not actually going to read it? > On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed

Re: [asterisk-users] Little t38 bug?

On 05/25/2010 07:54 PM, Kevin P. Fleming wrote: > On 05/25/2010 05:48 AM, Alexandru Oniciuc wrote: > >> Hello List, >> >> >> >> I think I’ve discovered a little bug in t.38 bug in >> 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. >> >> >> >>

Re: [asterisk-users] Need fax solution for 1.4.xx (Resolution)

On 05/13/2010 10:48 PM, William Stillwell (Lists) wrote: > Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp > 0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax. > > Hint: you need to install spandsp then run ./configure then make menuselect > :) > > > I was a

Re: [asterisk-users] Need fax solution for 1.4.xx

On 05/12/2010 08:46 AM, David Backeberg wrote: > On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists) > wrote: > >> Anybody know a reliable fax solution for 1.4.30 branch? >> >> >> I am using PikaFax on another server and works very well (about 3000 faxes >> a week), but it appears the

Re: [asterisk-users] Questions About Fax for Asterisk

On 05/08/2010 08:15 AM, Steve Totaro wrote: > > > On Fri, May 7, 2010 at 2:01 PM, Martin > wrote: > > On Thu, May 6, 2010 at 3:11 PM, Steve Totaro > > wrote: > > Yes, I purchased licenses for Fax for Asteris

Re: [asterisk-users] ATA shootout: PAP2T versus Grandstream Handytone 286

On 05/02/2010 02:59 AM, James Lamanna wrote: >>> It seems that the PAP2T does support TFTP and an XML-based config for >>> deployments... >>> >>> > I've used both the Grandstream 286 and the Linksys PAP2T. > I have been able to get some limited faxing to work using T30 with a PAP2T. > Confi

Re: [asterisk-users] E3 Card on Asterisk ?

On 04/27/2010 10:41 PM, Anita Hall wrote: > Hi > > Please check out this product > > http://www.sangoma.com/products/hardware_products/data_networking/a301.html > > Does it work on Asterisk or Freeswitch ? > Do Telcos provide an E3 connection ? > > One of our customers had an inquiry for terminatin

Re: [asterisk-users] fax & spandsp

On 03/09/2010 07:31 AM, Edwin Lam wrote: > hi folks. > > i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having > problems with fax. after receiving fax with the ReceiveFAX app. > everything seems ok. the .tiff file was there, phone line seems > to hang up. then asterisk will crash. any

Re: [asterisk-users] t38 ATA

On 03/13/2010 02:03 AM, Jeff Brower wrote: > Steve- > > >> On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: >> >>> Hello, >>> >>> I need a hand in choosing a small ATA, even with one FXS port, that >>> should do only fax with T38. >>> >>> I’ve tried Grandstream (ht286 model) but the faxes

Re: [asterisk-users] t38 ATA

On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote: > > Hello, > > I need a hand in choosing a small ATA, even with one FXS port, that > should do only fax with T38. > > I’ve tried Grandstream (ht286 model) but the faxes go out without ECM, > even if the Fax machine has ECM enabled. > > Is there any

Re: [asterisk-users] Calculating R Factor and MOS metrics for VoIP

On 03/09/2010 12:30 AM, Kevin P. Fleming wrote: > Steve Underwood wrote: > > >>> MOS and R factor are the two QoS parameters used to estimate VoIP call >>> quality. >>> >> You can't calculate MOS. Its an assessment based on a lot of human

Re: [asterisk-users] Calculating R Factor and MOS metrics for VoIP

On 03/08/2010 06:10 PM, mosbah.abdelkader wrote: > Hello All, > > > > MOS and R factor are the two QoS parameters used to estimate VoIP call > quality. You can't calculate MOS. Its an assessment based on a lot of human hearing. Nonetheless, there is a profitable industry in pretending to calcula

Re: [asterisk-users] SIP / Echo Cancellation

On 03/05/2010 02:45 PM, Vineet Bhojnagarwala wrote: > Very informative post Vinícius ! > > 2010/3/5 Vinícius Fontes > > > - "Chandrakant Solanki" > escreveu: > > > Hello > > > > I have successfully compil

Re: [asterisk-users] Slightly OT: Has SILK codec gotten anywhere?

On 02/21/2010 02:02 AM, Kyle Kienapfel wrote: > Hi, I stumbled upon mentions of a "SILK" codec last night on skypes > "skype for sip" information page. I tried looking into it further and > found some blog and mailing list posts from 2009 but I can't find any > mentions of anything other than skyp

Re: [asterisk-users] Asterisk t38modem Fax gateway evaluation

On 02/18/2010 03:40 PM, dle...@lstelcom.com wrote: > Hi, > > I am trying to fix a Asterisk setup with buggy (POTS) Fax machines. The > setup consists of the following components: > > - A Digium TE121 for connectiong to E1 ISDN > - Debian box with Asterisk 1.4 > - Grandstream GXW-4008 SIP ATA to whi

Re: [asterisk-users] Still on spandsp/app_fax and T.38

On 02/15/2010 11:27 PM, Kevin P. Fleming wrote: > Steve Underwood wrote: > > >> FFA sends its repeating no-signal and preamble packets with incrementing >> sequence numbers. While its not the only system which does that, it >> confuses some T.38 implementations. Th

Re: [asterisk-users] Still on spandsp/app_fax and T.38

t;> goes okay with that change, I'll post a patch on Mantis. >>> >>> No need for the patch; it's already on my radar, and if you confirm >>> that >>> it actually solves an interop problem, I'll commit the update to >>> >&

Re: [asterisk-users] Still on spandsp/app_fax and T.38

t; > > Atenciosamente, > > Vinícius Fontes > Gerente de Segurança da Informação > Canall Tecnologia em Comunicações > Passo Fundo - RS - Brasil > +55 54 2104-7000 > > Information Security Manager > Canall Tecnologia em Comunicações > Passo Fundo - RS - Brazil

Re: [asterisk-users] Still on spandsp/app_fax and T.38

On 02/05/2010 07:40 PM, Vinícius Fontes wrote: > 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 d

Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing

On 02/03/2010 03:14 AM, Kevin P. Fleming wrote: > Steve Underwood wrote: > > >> I wonder why Asterisk would say: >> >> X-asterisk-Info: SIP re-invite (External RTP bridge) >> Content-Type: application/sdp >> Content-Length: 344 >> >> v=0 &g

Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing

On 02/03/2010 12:45 AM, Vinícius Fontes wrote: > - "Kevin P. Fleming" escreveu: > > >> Vinícius Fontes wrote: >> >> >>> I couldn't agree more Steve. >>> >>> Is there any other info I could provide in order to help you find >>> >> out what's wrong? I could even open an issue o

Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing

On 02/02/2010 10:11 PM, Kevin P. Fleming wrote: > Steve Underwood wrote: > >> Hi Kevin, >> >> On 02/02/2010 09:12 PM, Kevin P. Fleming wrote: >> >>> Vinícius Fontes wrote: >>> >>> >>> >>>> [Feb 2 08:38

Re: [asterisk-users] Asterisk 1.6.1.13 and T.38 faxing

Hi Kevin, On 02/02/2010 09:12 PM, Kevin P. Fleming wrote: > Vinícius Fontes wrote: > > >> [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing >> session-level SDP v=0... UNSUPPORTED. >> [Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing >> session-lev

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

On 01/17/2010 04:11 PM, Doug wrote: > At 23:04 1/16/2010, Tilghman Lesher wrote: > > >That's incorrect. "module show" shows only those modules which are > currently > >loaded. BTW, there is also the command "module show like fax", which is > much > >easier than typing out the whole module

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

On 01/17/2010 06:10 PM, IT-Connect wrote: > Hallo there! > I had my own experience get RxFax/TxFax successful running with spandsp. > I only got spandsp-0.0.4 running, because on newer package, there > aren't created some needed libraries (don't remember the right one > this moment) Spandsp only

Re: [asterisk-users] Faxing: Anyone have a compiled executable?

On 01/08/2010 06:05 AM, William Stillwell (Lists) wrote: > Has there been any improvement with app_fax ? > > I stopped using it as I had a high failure rate with inbound faxes (10%+) > 1000 faxes a week ,with over a 100 failures can get quite annoying from > people complaining.. I could get it to f

Re: [asterisk-users] iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P

On 01/07/2010 07:21 PM, Olivier wrote: > PS: If you compile Asterisk from source after installing spandsp, > SendFAX and ReceiveFAX would automatically be included. > I opened another thread about that but I doubt that both SendFAX and > ReceiveFAX behave exactly the same (no side effect), no mat

Re: [asterisk-users] rxgain / txgain for iaxmodem or hylafax

On 12/27/2009 10:10 PM, Barry Fawthrop wrote: > In trying to get the asterisk and faxing working > I had to resolve to using iaxmodem and hylafax. > I have incoming working, but outgoing the other fax rings > but it would appear from web searches that the fax signals > are too low to be "heard" >

Re: [asterisk-users] Asterisk 1.6.1.11 Fax

On 12/11/2009 03:33 AM, Kevin P. Fleming wrote: > Cyprus VoIP wrote: > > >> Before posting my question, I analyzed the entire SIP negotiation and it >> was fine. The problem began in the T.38 negotiation itself, after the >> Asterisk's reINVITE. I don't have the old calls traces anymore, but I'

Re: [asterisk-users] G729: TC400B vs Software Encoder

On 12/05/2009 10:19 PM, Kevin P. Fleming wrote: > Jim Boykin wrote: > >> Can someone help me decide between TC400B vs Software Encoder. TC400B >> is really expensive and would like to get opinion if it is worth >> considering. Maximum simultaneous call we handle is hardly 10-15. Will >> going t

Re: [asterisk-users] spandsp version

On 12/04/2009 06:54 PM, Magnus Benngård wrote: > Hi! > > What version of spandsp is recommended to use when u compile > asterisk-trunk? The next one, or if that hasn't been released yet, the current one. Steve ___ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Location

I thought the internet was entirely driven by the negative energy of its users. Monsters Inc was based on the Internet, wasn't it? Steve On 11/07/2009 08:31 AM, Thomas Perron wrote: > I am trying to find others in my area. > Have a sense of enjoyment instead of a negative attitude. > > > On Fri

Re: [asterisk-users] Asterisk and Software Data Modem

On 11/03/2009 09:56 PM, Cherif wrote: > > Hello everybody > > I am trying to connect my asterisk to a payment equipment trough PSTN. > > I have a TDM400P card with an fxs module an the equipment use modem to > send data! > > I was thinking to implement a software data modem in asterisk, but I > f

Re: [asterisk-users] Asterisk 1.4 and Fax

On 11/03/2009 04:25 AM, Thomas Kenyon wrote: > Kevin P. Fleming wrote: > >> Lee Howard wrote: >> >>> I've heard of people who go to casinos and come home with a couple >>> thousand bucks winnings, too. But the truth is that invariably the vast >>> majority of people who gamble don't win.

Re: [asterisk-users] What are the reasons for VoIP echo?

On 10/03/2009 02:18 AM, Martin wrote: >> You might like to know that a number of people have used the open source >> OSLEC canceller to replace the rather broken one Linksys put in their ATAs. >> > there's no way to cancel the sip device in Asterisk, is there ? other > than going through zap

Re: [asterisk-users] What are the reasons for VoIP echo?

On 10/02/2009 09:18 AM, Martin wrote: > Are you saying there are half duplex phones out there with half > duplex speakerphones ? > Practically all analogue speakerphones are half duplex. Only a small number of analogue phones ever implemented a proper echo canceller based speakerphone -

Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed

On 10/02/2009 01:44 AM, Martin wrote: > anyone can just grab the PEF framer datasheet and tweak the driver though... > last I checked there's a whole section devoted to high impedance in > the datasheet > > Martin > The hardware needs to have been built in a particular way, with particular ver

Re: [asterisk-users] What are the reasons for VoIP echo?

On 10/02/2009 08:36 AM, Martin wrote: > if a user calling you hears echo of himself then it's the fault of > your sip device/sip phone. > The manufacturer must be using a cheap or an open source echo canceller ... > > try getting a different sip device made by some 'normal' company like > polycom o

Re: [asterisk-users] DTMF end '1' has duration 57 but want minimum 80 , emulating on ?IAX2/a16-q1-9657

On 09/21/2009 12:46 PM, Tilghman Lesher wrote: > On Sunday 20 September 2009 22:32:41 Tzafrir Cohen wrote: > >> Isn't 40ms the minimal time for a valid dtmf digit? >> > $ grep -C1 'define AST_MIN_DTMF_DURATION' main/channel.c > /*! Minimum allowed digit length - 80ms */ > #define AST_MIN_

Re: [asterisk-users] G729

On 09/17/2009 02:52 PM, Gordon Henderson wrote: > On Wed, 16 Sep 2009, Tilghman Lesher wrote: > > >> On Wednesday 16 September 2009 06:46:13 Khaled W Chehab wrote: >> >>> What g729 module should I download ? >>> >> You should download only the licensed g.729 module from Digium, a

Re: [asterisk-users] HPEC > VPM ?

Noah Miller wrote: >>> Next question: does anybody know how to handle extremely long tail >>> echo that a VPM module cannot? >>> >> How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can >> handle 128ms echo tails, which is pretty darn long. It's rare to see an >> echo tail longe

Re: [asterisk-users] OT - Do analog gateways detect a phone is plugged in or out ?

Lyle Giese wrote: > Philipp Kempgen wrote: >> Steve Totaro schrieb: >> >>> On Wed, Jul 22, 2009 at 12:07 PM, Steve Underwood wrote: >>> >>>> Olivier wrote: >>>> >>>>> I've got a general question about an

Re: [asterisk-users] OT - Do analog gateways detect a phone is plugged in or out ?

Olivier wrote: > Hi, > > I've got a general question about analog gateways (Xorcom, Audiocodes, > Patton, ...) . > Is it usual for analog gateways to detect when an analog phone is > plugged in or out ? > If positive, would it be then useful to send "qualify" queries for > each connect phone (I'

Re: [asterisk-users] T38 negotiation, the last step !

Klaus Darilion wrote: > Xavier Cardil schrieb: > >> Hi, I've managed to get HYLAFAX>T38MODEM-> >> ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk >> drops a message telling "Unknown RTP codec 96 received from gateway" Do >> somebody know how to fix it ? >> >>

Re: [asterisk-users] g.722 + loudness

Hose wrote: > What you say...Dave Fullerton (dfullertaster...@shorelinecontainer.com): > > >> Kevin P. Fleming wrote: >> >>> Hose wrote: >>> >>> I have a feeling that the issue is between transcoding of ulaw to g.722 and it's too loud during the transcoding - anyway to adju

Re: [asterisk-users] g.722 + loudness

Hose wrote: > Hi, > > > >

Re: [asterisk-users] T.38 pass-through 488 handling problem

Klaus Darilion wrote: > Steve Underwood schrieb: > >> >> There seems to be a common misconception about 488. It represents an >> irrevocable failure of the call. Once a 488 is sent the call is >> essentially dead. A number of systems are able to contin

Re: [asterisk-users] T.38 pass-through 488 handling problem

Klaus Darilion wrote: > Atis Lezdins schrieb: > >> On Mon, Jun 8, 2009 at 2:06 PM, Klaus >> Darilion wrote: >> >>> Hi! >>> >>> I have the following problem with Asterisk 1.4.23: >>> >>> >>> ATA w/ T.38 Asterisk ATA w/o T.38 >>> INVITE> >>>

Re: [asterisk-users] Digium Fax Driver

rovided excellent support, there >>> never would have been a reason for IAXmodem to be developed. >>> >> Reminder: SpanDSP was originally GPL. Steve Underwood wrote rx_fax.c and >> tx_fax.c that were basically thin wrappers for it. They were not so easy >>

Re: [asterisk-users] Digium Fax Driver

Lee Howard wrote: > Tilghman Lesher wrote: > >> On Sunday 07 June 2009 19:39:50 Lee Howard wrote: >> >> >>> Tilghman Lesher wrote: >>> >>> > What's the use case for the Digium > driver? Am I missing something by not using it? > > While

Re: [asterisk-users] Digium Fax Driver

Elliot Murdock wrote: > Hello! > I have a 64 bit Asterisk system and am wondering how to use Digium's > 32 bit fax driver. Is there some kind of emulation that can be used? > Thanks! > Elliot Use the FAX support built into Asterisk 1.6 and you won't have that limitation. Steve ___

Re: [asterisk-users] [Atcom] Asterisk + LAMP on 128MB RAM?

Tzafrir Cohen wrote: > On Mon, Jun 01, 2009 at 11:27:53AM +0200, Vincent wrote: > >> Hello >> >> I'm thinking of selling an Asterisk server based on Atcom's IP02 >> solid-state unit with one FXO and one FXS ports: >> >> http://atcom.cn/En_products_IP02.htm >> >> By default, this unit based

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] 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] 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] 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

[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] 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

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] 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] 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] 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] 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] 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

[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] 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

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] 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] [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] [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?

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] 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] 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] 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] 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] 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] 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 ?

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] 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] 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] 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] 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] 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] 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] 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] 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

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] 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] 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] 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] 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] 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] 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] 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] 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] 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] How to set udptl.conf ?

Mark Michelson wrote: > Olivier wrote: > >> Hi, >> >> voip-info.org is almost silent regarding >> udptl.conf except with >> "Depending on your fax device (such as the Linksys 3102) you may have >> to edit the udptl.conf file. The error correction type that is sent is >

Re: [asterisk-users] Quiet 24 port POE gig switch

Singer XJ Wang wrote: > [snipped] >> You can do that by using fans other than the tiny, whiney, 40mm fans >> that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin >> fans at the back or front, pushing air in (hence the deep >> dimensions), but the top and bottom would need recesses to

Re: [asterisk-users] Quiet 24 port POE gig switch

Gordon Henderson wrote: > On Mon, 2 Feb 2009, Steve Underwood wrote: > > >> Bernd Felsche wrote: >> >>> Ian Cowley wrote: >>> >>> >>> >>>> Beware PoE switches that can't handle Class

Re: [asterisk-users] Quiet 24 port POE gig switch

Bernd Felsche wrote: > Ian Cowley wrote: > > >> Beware PoE switches that can't handle Class 3 (15W) on all ports. >> Most have fans because 24 (or 48) x 15W is hot! >> > > That's the power supplied .. which'd be at the far end of the wire. > > The efficiency of the PSU plays a big part in

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