Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?
On 03/30/2016 08:23 PM, Vitor Mazuco wrote: Hi! Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or any others digium card FXO for use Fax modem? Thanks. Asterisk + iaxmodem gives you a bunch of soft FAX modems. Add one of the analogue PSTN interface cards you listed and you have a multi-channel PSTN connected FAX modem. This arrangement is widely used with HylaFAX, although people do use it with other FAX software, such as the stuff built into Windows (using ethernet virtual terminals to connect the windows box to the linux box). Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
On 04/09/2014 06:54 PM, Tzafrir Cohen wrote: On Wed, Apr 09, 2014 at 10:19:59AM +0800, Steve Underwood wrote: Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. If it allows me to avoid the trolls: I'll pay that performance hit. In many caces there are CPU cycles to spare. But the licensing is a hard limit. Well, you do get the benefit of higher quality for your extra compute. G.729 sounds distinctly better than G.729A on a lot of material. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Nice. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
Hi Jeff, On 04/08/2014 12:13 PM, Jeff Brower wrote: Darrel- The G729 essential patents were *granted* in 1996, but applied for prior to June 8 1995. That means their lifespan is either 20 years from their application date, or 17 years from their grant date, whichever is greater (http://www.uspto.gov/main/faq/p120013.htm). Either way, they expire in 2014. -Jeff Where did you get the cutoff date of June 8 1995, and how does 20 years from that date lead to the last of the patents expiring in 2014? Nobody uses G.729. They use G.729A. The G.729A spec is somewhat later than the original G.729, but I don't know if there are any additional patents which specifically relate to Annex A. You could use G.729 instead, but it roughly doubles the compute needed. There are various things on the web saying the last of the patents on G.723.1, which was around in draft form long before G.729, expires in 2014. However, there seem to be patents related to that codec which don't really expire until some time in 2015. Its really hard to find solid information. The ITU patent database rarely identifies the actual patents being claimed, so its damned hard to look them up. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oddity with FFA
On 03/11/2014 12:36 AM, Mike Diehl wrote: Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look at the actual .tiff image, it looks to be complete. This is causing our users to not get a positive acknowledgement when they send the fax. Is there anything we can do to mitigate this? Mike. How do you know the FAX is complete? If a page was received, the sending machine said more pages were to follow, and then it dropped the call, is that a complete FAX? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oddity with FFA
Hi Mike, If the sending machine keeps trying it might be the call has been hung up by asterisk before its own acknowledgement message has finished being sent. There have been bugs like this in the past, and people can be pretty casual about making changes which hang up aggressively. A FAX system should really wait for the final DCN message before disconnecting, to ensure both sides have seen what they need. Spandsp does that, but I am not sure about FFA. Regards, Steve On 03/11/2014 03:03 AM, Mike Diehl wrote: Steve, I BELIEVE the fax is complete because the fax image is a form that appears to only be a single page. But, since FFA isn't providing acknowledgement, the sending fax machine is resending the document multiple times! Mike. On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood ste...@coppice.org mailto:ste...@coppice.org wrote: On 03/11/2014 12:36 AM, Mike Diehl wrote: Hi all, For the most part, we are finding that Fax for Asterisk works pretty well. However, we have seen some wierdness that we'd like to try to fix. Once in a while, we will get a partial result report for a given fax but when we look at the actual .tiff image, it looks to be complete. This is causing our users to not get a positive acknowledgement when they send the fax. Is there anything we can do to mitigate this? Mike. How do you know the FAX is complete? If a page was received, the sending machine said more pages were to follow, and then it dropped the call, is that a complete FAX? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
On 10/05/2013 11:07 PM, Darryl Moore wrote: On 2013-10-04 5:36 PM, Steve Underwood ste...@coppice.org mailto:ste...@coppice.org wrote: On 10/05/2013 01:32 AM, Darryl Moore wrote: I'll explain. The g.729 compression algorithm is not protected by copyright, though specific instances may be. It is protected by a patent. http://www.sipro.com/G-729.html An open source version is available here: http://asterisk.hosting.lv/ What stops you from using this, or even your own implementation isn't copyright, but patent protection. It is the right to use the patented technology that you are licensing, not the particular copyrighted coded that implements it. The G.729 codec software at http://asterisk.hosting.lv/actually uses a codec implementation copyrighted by Intel. You need to obey their copyright conditions. correct, and for a few hundred dollars you are free to use it as you see fit, without royalties. note that i also said that the patent license applies even on code that you write yourself. Here you will find the various G.729 patents which were all granted in 1996. https://www.itu.int/ITU-T/recommendations/related_ps.aspx?id_prod=3334 I had thought these expired next year because I was thinking patents were only 18 years. Turns out they are now 20 years, so they really do not expire til some time in 2016. My bad. If you use G.729A (which practically everyone does) I think there are one or two patent which run beyond 2016, at least in the US. perhaps. i do not claim to have fully researched either the patents or the protocol. is 729 compatible with 729a? out of curiosity though i will find out more about these other patents. So in countries that honour software patents, you need to have a license until some time in 2016. In countries which do not, you are free to use these open source codes now. What have the essential patents relevant to G.729 got to do with software patents? [blink] umm... they are software patents. Really? Do you have expert legal opinion on that? I've never seen anyone competent dispute the patentability of applied signal processing. Such patents get issued all over the world. There are a couple of software patents related to G.729, but those are not part of the essential pool of patents, and those are probably US only. cheers. On Fri, 2013-10-04 at 15:55 +0200, Olivier wrote: H, I'm not sure how g729 licence and software patents relate to each other. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
On 10/05/2013 01:32 AM, Darryl Moore wrote: I'll explain. The g.729 compression algorithm is not protected by copyright, though specific instances may be. It is protected by a patent. http://www.sipro.com/G-729.html An open source version is available here: http://asterisk.hosting.lv/ What stops you from using this, or even your own implementation isn't copyright, but patent protection. It is the right to use the patented technology that you are licensing, not the particular copyrighted coded that implements it. The G.729 codec software at http://asterisk.hosting.lv/actually uses a codec implementation copyrighted by Intel. You need to obey their copyright conditions. Here you will find the various G.729 patents which were all granted in 1996. https://www.itu.int/ITU-T/recommendations/related_ps.aspx?id_prod=3334 I had thought these expired next year because I was thinking patents were only 18 years. Turns out they are now 20 years, so they really do not expire til some time in 2016. My bad. If you use G.729A (which practically everyone does) I think there are one or two patent which run beyond 2016, at least in the US. So in countries that honour software patents, you need to have a license until some time in 2016. In countries which do not, you are free to use these open source codes now. What have the essential patents relevant to G.729 got to do with software patents? cheers. On Fri, 2013-10-04 at 15:55 +0200, Olivier wrote: H, I'm not sure how g729 licence and software patents relate to each other. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue in transcoding
On 06/02/2013 11:02 PM, Chris Bagnall wrote: On 2/6/13 2:01 pm, Muhammad Yousuf wrote: I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0 Isn't g723.1 considered pretty poor quality these days? Can't you set voipswitch to use something apart from that? For a 5.3kbps codec G.723.1 is still pretty good, and I don't know another codec with a similar bit rate that is available on as many platforms. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disagreements between codec_siren14 and Polycom sources
On 03/15/2013 10:41 AM, Richard Kenner wrote: There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back correctly with the Digium's siren14 codec. I know the parameters are correct because the file is the same size as that made by the Digium codec. Both sets of decoders/encoders (Digium and Polycom/ITU) are symmetric and can decode what they encode, but neither can read the encoding of the other. Is there some subtle difference between G.722.1C and Siren14? G,722.1C is not the same as Siren 14. This is stated in the Polycom material but they don't really indicate how different the two are. More importantly, they are vague about whether the two can be expected to interwork satisfactorily. Polycom only offer source code for G.722.1C, so you can't really figure out the differences for yourself. People are really sloppy about these names and something called siren14 might well be G.722.1C. I assume something called G.722.1C is always G.722.1C. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/09/2012 12:28 AM, Brett Lehrer wrote: How many fax and voice calls (which codecs for tha latter ones ?) are on average using your DSL line ? 1. Previously, I experienced failures during the process of converting incoming PDF documents into ready-to-send fax image files while the reverse process (from a fax file into a PDF or whatever document) never failed. I would be curious to check if a greater failure rate for outbound faxing (greater than inbound faxing failure rate) could simply comes from image processing, before any transmission. 2. Though your DSL line may have enough bandwidth from your location to its DSLAM, chances are packets are dropped or delivered too late for T.38 faxing. An interesting test would be to use an Asterisk PBX hosted somewhere at close range from netVortex fax gateways : that would remove most networking issues out of the equation. I'll have to look more closely into what codecs we traditionally use, but g.722 up and ulaw down is common. Generally don't have more than 2-3 calls active at once. At most, 5, and that's a rarity. Record for fax is 4 simultaneous send/receive, but typically just 1, maybe 2. I imagine that's encroaching on the upper limits of the 768 kbps upspeed. I've wondered about how lag might impact the problem but I just don't know how I'd go about testing it properly without spending a bunch of money on hosting. I do my PDF - TIFF conversion on another machine with ghostscript. Here's the line: gs -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg4 -sOutputFile=TIFF_FILENAME -f PDF_FILENAME I changed from tiffg3 to tiffg4 because the filesize got cut in half assuming that the less time spent transmitting, the less chance there was to run into a problem that might stop the fax. However, most failures that I've looked at seem to occur immediately or fail to connect at all, rather than get cut off due to a hiccup in the connection. Brett Lehrer A FAX can only be sent in ECM mode when using tiffg4 format. It will have to be recoded into tiffg3 format if ECM is inhibited, which it far too often is. On the other hand, if you are using ECM any decent FAX system (e.g. spandsp) will recode into tiffg4, and really good ones (e.g. the very latest spandsp) may recoed into T.85/JBIG, for faster transmission times. Digium don't seem to specify what FFA does in this area. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendFAX - multi-page TIFF
On 10/07/2012 04:56 PM, Mikhail Lischuk wrote: Gabriel Ortiz Lour писал 06.10.2012 17:07: I am using this command to generate the TIFF from a PDF: /usr/bin/gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 -dNOPAUSE -sOutputFile=$tiffFile -- $pdfFile I use imagemagic's convert instead of gs, for gs gave me lots of problems I had no time to debug: /usr/local/bin/convert -density 204x98 -resize 1728x1346 -units pixelsperinch -monochrome -compress Fax $timestamp.pdf $timestamp.tif And it works perfectly with multipage PDFs imagemagic just calls gs to do the actual work. The command you gave will not give you reliable conversion. It will work a lot of the time, to produce a low resolution FAX image. However, if you have large images in your PDF they can cause the output image size to change sometimes. The commands you will find at http://www.soft-switch.org/spandsp_faq/ar01s14.html will reliably produce standard, fine, and superfine image files in the various common FAX formats. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendFAX - multi-page TIFF
Hi Gabriel, There is something weird about your pages. They are supposed to be 1728x2292 pixels, and yet there are apparently 4969 rows in the strip. Regards, Steve On 10/06/2012 10:07 PM, Gabriel Ortiz Lour wrote: I am using this command to generate the TIFF from a PDF: /usr/bin/gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 -dNOPAUSE -sOutputFile=$tiffFile -- $pdfFile And tiffinfo for a 2 page generate file gives: # tiffinfo teste.tiff TIFF Directory at offset 0x8 (8) Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2292 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 4 Photometric Interpretation: min-is-white FillOrder: msb-to-lsb Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 4969 Planar Configuration: single image plane Page Number: 0-0 Software: GPL Ghostscript 8.71 DateTime: 2012:10:04 22:16:14 Group 4 Options: (0 = 0x0) TIFF Directory at offset 0x394 (916) Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 2292 Resolution: 204, 196 pixels/inch Bits/Sample: 1 Compression Scheme: CCITT Group 4 Photometric Interpretation: min-is-white FillOrder: msb-to-lsb Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: 4969 Planar Configuration: single image plane Page Number: 1-0 Software: GPL Ghostscript 8.71 DateTime: 2012:10:04 22:16:14 Group 4 Options: (0 = 0x0) I think this is correct, since it came from asterisk FAXing Howtos. Is that correct? I'll try doing some more tests with debug info ON and post back the results. Thanks, Gabriel 2012/10/5 Steve Underwood ste...@coppice.org mailto:ste...@coppice.org On 10/06/2012 02:53 AM, Gabriel Ortiz Lour wrote: 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 Check the file with tiffinfo. Perhaps the format of the second page is different from the first, and incompatible with faxing. Spandsp will renegotiate and continue sending if the second page is of a different but valid format, but sending will end if the second page is, say, colour or the wrong width. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendFAX - multi-page TIFF
On 10/06/2012 02:53 AM, Gabriel Ortiz Lour wrote: 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 Check the file with tiffinfo. Perhaps the format of the second page is different from the first, and incompatible with faxing. Spandsp will renegotiate and continue sending if the second page is of a different but valid format, but sending will end if the second page is, say, colour or the wrong width. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax for Asterisk success rates?
On 10/04/2012 09:29 PM, Brett Lehrer wrote: I'm running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VoIP usage. For both incoming and outgoing faxes, I'm getting a failure rate of just over 25%, and over a handful of reasons. Is it natural to have this many problems on a completely digital configuration? I'm trying to cut our analog phone line (because it's so expensive), but some fax machines just don't seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through. Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)? Brett Lehrer Unexplainable FAX call failures (i.e. not wrong numbers of other obviously wrong things) should be well below 1%. On a dedicated DSL line, if everything is set up properly you should be getting that kind of rate. This is especially true if you are using T.38 and the provider at the far end uses a decent T.38 platform. Across the open internet results are much more variable. Depending what causes your 25% failures, you may get better results with spandsp than with FFA. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM Fax
On 08/19/2012 11:45 AM, Lee Howard wrote: On 08/17/2012 04:58 AM, Steve Underwood wrote: On 08/17/2012 06:08 AM, Eric Wieling wrote: Has anyone experimented with increasing the DAHDI chunk size in improve fax reliability? If so, did it help, hurt, or not make any difference? I haven't found issues related to the DAHDI chunk size. The main thing which used to hurt FAXing with Asterisk before Digium launched their own FAX software was the timing within Asterisk, which they refused to fix at that time (although independent patches were available). With the launch of FFA they changed chan_dahdi so on a FAX call the buffering should change to make the flow of transmitted audio a lot more elastic. People just tolerate some hiccups in voice calls, but hate latency. Modem signals must be rigidly timed, but a bit more latency is OK. This change fixed the main issue affecting all the FAX solutions around. If that switch in the buffering mode is not happening on your system for some reason it can badly affect the reliability of FAXes. I'm uncertain of exactly to which changes you're referring. Your comments seem to fall in-line with the notion behind the DAHDI buffers feature for the channel as well as the DAHDI fax-detection faxbuffers feature, but I'm seeing no noticeable improvement, AND I'm uncertain how to implement the CHANNEL(buffers) feature due to: -- Executing [4628160@fax-outbound:1] Set(IAX2/ttyIAX99-584, CHANNEL(buffers)=12,half) in new stack [Aug 18 20:12:40] WARNING[6381]: func_channel.c:530 func_channel_write_real: Unknown or unavailable item requested: 'buffers' -- Executing [4628160@fax-outbound:2] Goto(IAX2/ttyIAX99-584, outbound,4628160,1) in new stack -- Goto (outbound,4628160,1) -- Executing [4628160@outbound:1] Dial(IAX2/ttyIAX99-584, DAHDI/g0/4628160) in new stack On some installations there are occasional instances in most outbound calls where Asterisk creates what otherwise would be considered jitter on the DAHDI channel. Generally these do not cause much real-world trouble, but I'm a stickler for perfect audio quality on all-digital calls. I've seen this on Asterisk versions 1.4, 1.6, and 1.8. On other installations there never is any such trouble noticeable. Would you mind being a bit more specific on the Asterisk changes to which you refer and how they should be implemented in the configuration? I was referring to the DADHI buffer control, which is (or was the last time I looked) tied in with the DADHI channel's fax detection scheme. It was never that big a problem with iaxmodem for some reason. The timing of things passing through Asterisk was always handled more smoothly than the timing of things originating from within Asterisk. For smooth audio flow you need to have plenty of audio buffered up in the DADHI transmit queue. Sometimes DADHI doesn't get serviced for quite a long time, and the queue needs enough stored audio to prevent underflows. The same issue can cause buffer overflows on the audio receive side, and additional buffers certainly help there, but underflow on the transmit side was always the dominant problem. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM Fax
On 08/17/2012 06:08 AM, Eric Wieling wrote: Has anyone experimented with increasing the DAHDI chunk size in improve fax reliability? If so, did it help, hurt, or not make any difference? I haven't found issues related to the DAHDI chunk size. The main thing which used to hurt FAXing with Asterisk before Digium launched their own FAX software was the timing within Asterisk, which they refused to fix at that time (although independent patches were available). With the launch of FFA they changed chan_dahdi so on a FAX call the buffering should change to make the flow of transmitted audio a lot more elastic. People just tolerate some hiccups in voice calls, but hate latency. Modem signals must be rigidly timed, but a bit more latency is OK. This change fixed the main issue affecting all the FAX solutions around. If that switch in the buffering mode is not happening on your system for some reason it can badly affect the reliability of FAXes. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] best free fax solution with asterisk
On 08/12/2012 10:32 AM, James Sharp wrote: On 8/11/2012 8:05 AM, virendra bhati wrote: Hi team, I want to configure fax with asterisk. there a lot of fax link i found by google but not working perfectly. my setup as follow asterisk 10.x centos 5.8 Want to used T.38 with SpanDSP... Please suggest me the best way. and how to test FoIP ? I use Asterisk 10.3.1, SpanDSP 0.0.6, and Ubuntu 11.10 connecting to Gafachi.com. It works with probably 95% success rate talking via T.38. 95% is pretty bad. Do you know if the failures are mostly during the initial negotiation, or somewhere in the actual FAX exchange? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
On 07/18/2012 09:43 PM, Matthew Jordan wrote:
- Original Message -
From: Alejandro Recarey a...@recarey.org
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Wednesday, July 18, 2012 6:30:26 AM
Subject: [asterisk-users] Using Asterisk 10.6 as a T38 Fax gateway
Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten = _X.,n,Dial(SIP/${EXTEN}@x.x.x.x)
The correct setting is not FAXOPT(t38gateway) - that is not a valid parameter
to pass to the FAXOPT function. As you mention below, the correct setting
is Set(FAXOPT(gateway)=yes). The optional timeout is fine.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_FAXOPT
I have tried with both FAXOPT(t38gateway) and FAXOPT(gateway). I have
also tried setting t38pt_udptl = yes,redundancy in sip.conf. None of
these things work. When we send a fax:
1. Asterisk does NOT send a REINVITE with the t38 offered. Reading
the
documentation, it should detect the fax tone with the audiohook and
then send a REINVITE with t38 capability.
Have you confirmed that Asterisk does not send the re-INVITE using either
a packet sniffer or by monitoring the log with 'sip set debug on'? Without
seeing the SIP message traffic and a DEBUG log, its hard to say what
might be the cause of your issues.
Typically, I would expect to see something like the following in a DEBUG log:
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: detected v21 preamble from
SIP/ast1-g711-0001
[Jul 18 08:29:18] DEBUG[20234] res_fax.c: requesting T.38 for gateway session
for SIP/ast1-t38-
Note that this also answers your question in a subsequent e-mail: you
should be using res_fax, with either res_fax_spandsp or Fax for Asterisk.
2. Asterisk does not offer t38 in the SDP of the initial INVITE. This
is not a problem if it correctly detects and REINVITES for faxes, but
our destination carriers tell us that they cannot do the REINVITE
themselves because we do not offer t38 in our SDP, so they believe we
do not have that capability.
Obviously I would prefer to just detect the fax myself and have
asterisk do the REINVITE.
I have read all of the documentation on the asterisk wiki (which is
rather short) and anything else I could find online. Unfortunately
most of it is out of date and refers to asterisk versions 1.4 to 1.8,
which do not have T38 Gateway capability.
There typically isn't a lot of configuration that is needed for T.38
gateway support. The necessary dialplan configuration is documented
here:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
One thing that page doesn't mention is only spandsp supports T.38
gateway right now. The Digium FAX module does not.
Regards,
Steve
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Re: [asterisk-users] SendFAX timestamp
Hi David, The old app_fax code, which allowed spandsp to be used with Asterisk before Digium introduced the new modules supported the features you want. Maybe someone can go through that code and port the feature into the current res-fax code. Steve On 07/03/2012 09:57 AM, David Cunningham wrote: Kevin, Thanks for the reply. On 29 June 2012 00:29, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: On 06/27/2012 09:30 PM, David Cunningham wrote: Would anyone else know if Asterisk allows use of SpanDSP's time zone conversion? No, SendFAX (in res_fax) doesn't currently offer the ability to do what you are asking about. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low success rate for ReceiveFax
-- FAX handle 0: [ 016.274071 ], P30EVN_DOC_END -- FAX handle 0: [ 016.274086 ], STAT_FRM_MCF Channel 'DAHDI/i1/-4' fax session '0', [ 016.340258 ], channel sent 74 frames (1480 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 016.434711 ], stack sent 283 frames (5660 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 016.480280 ], channel sent 7 frames (140 ms) of silence. -- FAX handle 0: [ 017.459594 ], STAT_EVT_TX_V21_DONE st: F_END_ECM rt: FECMNFCS Channel 'DAHDI/i1/-4' fax session '0', [ 017.793706 ], stack sent 68 frames (1360 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 017.820355 ], channel sent 67 frames (1340 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 018.020382 ], channel sent 10 frames (200 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 019.400471 ], channel sent 69 frames (1380 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 019.420496 ], channel sent 1 frames (20 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 019.460489 ], channel sent 2 frames (40 ms) of energy. Channel 'DAHDI/i1/-4' fax session '0', [ 019.780498 ], channel sent 16 frames (320 ms) of silence. Channel 'DAHDI/i1/-4' fax session '0', [ 019.820519 ], channel sent 2 frames (40 ms) of energy. -- Span 1: Channel 0/7 got hangup request, cause 16 -- FAX handle 0: [ 019.879779 ], STAT_EVT_TMR_INT_EXP st: F_END_ECM rt: NTIX -- FAX handle 0: [ 022.382966 ], STAT_EVT_HW_CLOSE st: WT_HW_CLS rt: WCLSNCLS -- FAX handle 0: [ 022.383063 ], STAT_SES_COMPLETE -- FAX handle 0: [ 022.383083 ], P30EVN_COMPLETE == Spawn extension (fax-rx, receive, 19) exited non-zero on 'DAHDI/i1/-4' On Fri, Jun 22, 2012 at 12:25 PM, Steve Underwood ste...@coppice.org wrote: On 06/22/2012 11:58 AM, Roi Stork wrote: Hi, Im able to send faxes with no errors, but the success rate for the receiving side is less than 50%. Asterisk usually returns records these errors as partial fax and fax protocol error. A lot of the error values returned by FAXOPT are 3RD_T2_TIMEOUT and T2_TIMEOUT. Any suggestions on how to improve the fax receiving rate? I have a problem. Can you fix it? is not really a meaningful question. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SendFAX timestamp
On 06/26/2012 10:24 AM, David Cunningham wrote: Hello, Does SendFAX have the ability to put the caller ID and timestamp on the fax? If so, is there a way to adjust the timezone used for the timestamp? Thanks for any assistance. SpanDSP has that ability, including per instance time zones, but I don't know if the Asterisk module exposes that facility. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spandsp supports T.38?
On 06/22/2012 12:49 AM, Ahmed Munir wrote: Hi, I would like to know whether SpanDSP supports T.38 for Asterisk 10? Because as far as using Fax for Asterisk, I'm getting some issues using T.38 Only spandsp fully supports T.38 in Asterisk 10. The Digium module cannot work in gateway mode. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low success rate for ReceiveFax
On 06/22/2012 11:58 AM, Roi Stork wrote: Hi, Im able to send faxes with no errors, but the success rate for the receiving side is less than 50%. Asterisk usually returns records these errors as partial fax and fax protocol error. A lot of the error values returned by FAXOPT are 3RD_T2_TIMEOUT and T2_TIMEOUT. Any suggestions on how to improve the fax receiving rate? I have a problem. Can you fix it? is not really a meaningful question. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
On 05/17/2012 02:47 PM, gincantalupo wrote: Hi Steve, you are telling me there is no way to set a particular speed on my iaxmodem in order to force the sender speed? I have some problems with a customer who gets malformed faxes even if no error occurs. Since I cannot tell the sender to lower its fax speed, my idea is to force my iaxmodem to a lower fixed speed so the sender is oblidged to negotiate at that speed (or lower, of course) without the customer could realize it, at least at first. :) There is no ATA in the middle (I'm using it for my tests but my customer does not have any), all faxes are received thru a primary channel to a bunch of iaxmodems. Sometimes some faxes are corrupted, that's why I thought to lower the speed. I could try to disable ECM but that's even harder to do (found nothing on internet). You have a broken installation, and your response is to try to break it even more. Does that make sense? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problem on direct FXO port
Hi Sebastian, has still some issues that not all faxes pass ok, but does the work == still badly broken Your log doesn't seem to show a spandsp error. It looks more like a bad signal. Did you change anything else when you installed FFA? Usually people move the other way to improve their results. Steve On 05/18/2012 09:38 AM, Sebastian Gutierrez wrote: Rusty, thanks for the reply, the issue seems a spandsp issue, I changed to digium free asterisk fax and works much better, has still some issues that not all faxes pass ok, but does the work. thanks! On May 17, 2012, at 1:06 PM, Rusty Newton wrote: Sebastian, Seeing as this an issue related to faxing using the SpanDSP library; if you do not get an answer leading to a solution here, then you may try asking on the SpanDSP mailing list http://lists.soft-switch.org/mailman/listinfo It's likely that the Asterisk users, specifically using SpanDSP, may be on that list. Thanks, Rusty Newton Open Source Community Support Manager Digium, Inc |www.digium.com |www.asterisk.org On 5/16/2012 12:44 PM, Sebastian Gutierrez wrote: Hi, I´m with asterisk 1.6.2.20 DAHDI Version: 2.5.0.2 Echo Canceller: HWEC, MG2 SpanDSP: spandsp-0.0.6pre20.tgz http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre20.tgz FXO lines. Sending faxes works ok. but receiving gives me error: here is the debug: http://pastebin.com/qfFeXWQW any idea?? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
Hi, On 05/16/2012 09:59 PM, Larry Moore wrote: Read the subject line more closely. Tested receiving too, I set the Send Receive speed of the receiving analogue modem to that below, the log file on the sending modem (iaxmodem) reported it capable of 9600. May 16 21:32:04.28: [ 2335]: REMOTE best rate 9600 bit/s May 16 21:32:04.28: [ 2335]: REMOTE max A3 page width (303 mm) May 16 21:32:04.28: [ 2335]: REMOTE max unlimited page length May 16 21:32:04.28: [ 2335]: REMOTE best vres R16 x 15.4 line/mm May 16 21:32:04.28: [ 2335]: REMOTE format support: MH, MR, MMR May 16 21:32:04.28: [ 2335]: REMOTE supports T.30 Annex A, 256-byte ECM May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline May 16 21:32:04.28: [ 2335]: USE 9600 bit/s Perhaps the issue is with Hylafax. Setting the Transmit Receive strings to !24,48,72,96 seems to yield the most reliability in transmission If you have an ATA in the path that is often the case. Many of them badly mess up a FAX signal. Without such a distortion machine V.17 should be fine. Cheers, Larry. On 16/05/2012 7:23 PM, Larry Moore wrote: I have iaxmodem version 1.2.0 installed on my system. I have set the following in the IAX configuration file, SIGHUP'd FaxGetty and submitted a single page outbound fax via Asterisk; Class1RMQueryCmd: !24,48,72 # enable this to disable V.17 receiving Class1TMQueryCmd: !24,48,72 # enable this to disable V.17 sending The resulting output from my T.38 Gateway reports the following; -- Connection Statistics Bit Rate :7200 ECM : No Pages : 1 -- Hungup 'IAX2/iaxmodem0-11055' I also tested with the maximum speed set to 4800, the image was received however the responses to EOP timed out, I don't know if the is to do with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The result was the fax was retried for the defined number of attempts. Cheers, Larry. On 16/05/2012 6:28 PM, gincantalupo wrote: Hi all, I'm trying to lower my iaxmodem speed but still I haven't found any solution...I tried to add Class1RMQueryCmd: !24,48,72 to config.IAXtty but does not work...Hylafax says it it running at 9600 (sometimes at 14400) baud.. This is correct behaviour. The sending side has fine control over the modem modes it uses. The receiving side can only specify that V.27ter, or V.27ter+V.29 or V.27ter+V.29+V.17 are OK. So, if you allow the 7200bps mode of V.29 you are compelled to allows the 9600bps mode too. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax
On 05/03/2012 10:35 PM, cjwstudios wrote: If you're going full time hosted fax you will ultimately end up buying a t.38/sip gateway like an Audiocodes Mediant. Many people handling hundreds of thousands of FAXes per day would disagree with that assessment. On Thu, May 3, 2012 at 5:27 AM, Anita Hallanita.h...@simmortel.com wrote: Hi We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the results make us sad :( I suppose Asterisk also has the option of using spandsp or a commercial version from Commetrex. What are your experiences with receiving Fax on spandsp or commetrex on Asterisk ? Does it really matter whether I use Asterisk or FreeSWITCH ? regards, Anita Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E M signalling and Dahdi
On 04/20/2012 11:30 PM, Eduardo Pimenta wrote: Hello all, Does anyone know if EM over E1 signalling works on top of R2, ISDN and where can I find a sample Dahdi configuration? Have done a lot of google and cannot find a proper E1 configuration. No it doesn't. EM signalling is the same layer as R2 and ISDN. It is an alternative to them, not another layer. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding degradation G711-iLBC
On 04/15/2012 07:26 PM, Patrick Lists wrote: On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote: Is it a good idea to use asterisk transcoding from G711 to iLBC or should I find out any other solution not involving transcoding (f.e. using G.729 that is supported in both sides). I'm worried about voice quality and trying to avoid paying for G.729 licensing. Anybody with experience or quantitative measurements of the voice quality degradation in that scenario? The term that may interest you is Mean Opinion Score and iLBC is quite good. See http://en.wikipedia.org/wiki/Mean_opinion_score There's lies, damn lies and mean opinion scores. The chart on that wikipedia page is mostly for humour value. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 troubles
Hi Jean-Denis,
Your log shows the Mediatrix GW has problems. It sends a DCS signal to
the Asterisk box, but doesn't following it with TCF as it should. The
asterisk box times out waiting for TCF and tries to take recovery action
which fails.
Spandsp has some workarounds for bugs in Mediatrix boxes. They usually
work OK.
Regards,
Steve
On 03/27/2012 08:02 AM, Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
- -BEGIN PGP SIGNED MESSAGE-
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Hi list,
I'm having difficulties when receiving faxes from the PSTN with this
relatively simple installation:
PSTN--PRI-- GW--T.38-- Asterisk
The gateway is a Mediatrix 3301 (firmware Dgw 2.0.14.251). It's
configured to transmit faxes as T.38. I may have missed something in its
configuration, but it does switch to T.38 when a fax is detected. On the
Asterisk side, I'm using 10.2.1 with spandsp-0.0.6-pre20 and ReceiveFax
from res_fax_spandsp. ${FAXSTATUS} returns FAILED and ${FAXERROR}
Disconnected after permitted retries.
I did a network capture, attached to this mail: from my understanding,
T.38 is accepted by Asterisk, then there seems to be some UDPTL traffic,
which I don't understand...
Why does it fail, and what is wrong? I'd appreciate if someone could
send me advice / suggestions.
Thanks,
- - --
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française
http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27
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Re: [asterisk-users] outbound fax over t38 gateway can't pass
On 02/29/2012 02:28 PM, Dmitry Melekhov wrote: btw, played with res_fax.conf if I set maxrate=7200 fax machines try (and fail) 9600 anyway. Why? If limited ti 7200? looks like bug... Why do you think everything you don't understand is a bug? What you see is correct behaviour. Any party in the FAX chain can block V.17, or V.17+V.29. Only the entity sending a FAX can block individual modes. That's just how the FAX protocol works. So I set maxrate=4800 and modems=v27. Faxes pass Looks like problems with V29... I told you before what where the problem lies. It won't change by posting more messages like this. 29.02.2012 07:56, Dmitry Melekhov пишет: Hello! I have problems with outbound faxes with asterisk 10.2 t38 gateway. There is asterisk box, connected to panasonic kx-td500 over PRI link with TE122. If we try to send fax with following path: panasonic 500 extension fax machine panasonic500- asterisk- ooh323- cisco 3845- fax machine fax can't pass. always reproducable. as I see in tcpdump produced dump fax machines tries to connect on 9600 and failed, no attempt to down speed. If I send fax in path panasonic 500 extension fax machine - asterisk (ReceiveFAX) it is received successefully all the time. If I send fax from asterisk with SendFax as following: asterisk(SendFax) - panasonic500-asterisk- ooh323- cisco 3845...- fax machine it always passes. Usually on 7200, sometimes on 4800. So ooh323 works OK, fax part works OK, t38 works OK, but not with fax machine (we tested to different). Inbound faxes in reverse direction, i.e. fax machine...cisco3845- ooh323 - asterisk - panasonic - fax machine always pass on 7200. More info is here https://issues.asterisk.org/jira/browse/ASTERISK-19436 Bug report was closed because not a bug :-) Could you help me solve this problem? Thank you! Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
On 01/11/2012 02:39 PM, Olivier wrote: Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. As we all know, all updates are intended to break things, and the first rough draft of any package is the greatest perfection it will ever achieve. So, the question is how much destruction was caused in the update from pre17 to pre18? Probably a lot. I just broke things a bit more by posting pre19. This one breaks things by fixing a vulnerability in the FAX decompressor. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] local channels and g729a voice quality
On 01/16/2012 03:59 PM, Roi Stork wrote: Hi, We noticed a very sharp drop in voice quality when using digium g729a codec. The problem seems to happen if the A channel (caller's channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listening to a mono speaker on low volume. You have conflated two very different things there - landline calls and cellular calls. A land line to a VoIP user by G.729A should sounds pretty good. A cellphone to a VoIP user by G.729A should sound *far* worse. Converting between two different low bit rate codecs really hits the quality, and all cellphone calls are low bit rate. If I use a softphone that is directly registered to our asterisk box the audio quality improves, the words come out more clearer and louder. You are conflating two things again. Quality and volume are largely independent issues. I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. Try that again. If you really can't hear the difference you should check carefully that the system is working as you think it is. If it is, maybe you should consult a doctor. G.729A is considerably poorer than ulaw. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Installation in Asterisk
On 01/13/2012 05:17 PM, mahesh katta wrote: On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels ruben.roeg...@jumping-frog.org mailto:ruben.roeg...@jumping-frog.org wrote: Am 12.01.2012 18:50, schrieb mahesh katta: I was search for free license but for this Digium require purchase any Hardware then they can provide Free License. But I have no Digium Device , I am using Grand stream FXO Gateway and Asterisk.1.8.XX . I was connected like PSTN==FXOGateway==Asterisk(FXO configure through IP) If anything wrong please correct me. Hi Mahesh, the FreeFax for asterisk is really free and not bound to digium hardware, but it is limited to one concurrent fax session. At least you should be able to try if fax receiving is possible with this setup. As far as I can see, it should work with your setup. The URL I posted leads you to the FreeFAX for Asterisk Module. Sir,Its done.I receive the FAX.Thank you sir. One more thing sir if I sent at a time multiple fax to this is it receive. can you clarify me. scenario is I have PRI line of 30 channels. one Boarding no. if I send this is it receive the fax at a time with single free license. best regards, Ruben Remove the Digium FAX module and install SpanDSP. Then the number of FAXes you can receive at once will only be limited by the speed of your hardware. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a reliable T.38 Solution
On 01/11/2012 03:01 PM, Olivier wrote: 2012/1/5, Kevin P. Flemingkpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: 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! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? Are you really desperate to pay for functionality you can get for free? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a reliable T.38 Solution
On 01/11/2012 11:16 PM, Olivier wrote: 2012/1/11, Steve Underwoodste...@coppice.org: On 01/11/2012 03:01 PM, Olivier wrote: 2012/1/5, Kevin P. Flemingkpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: 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! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? Are you really desperate to pay for functionality you can get for free? Not yet ;-))) But the increased fax sending speed (14.4 kbs/s says the datasheet but I must be too naive to still read datasheets) may be a feature interesting for some. By the way, which spandsp version would recommend for asterisk 10 ? spandsp-0.0.6pre18.tgz ? How is 14.4k an increase? Both spandsp and the Digium modules do 14.4k. There is nothing the Digium module does which spandsp does not do, and the file handling in spandsp is more flexible. spandsp-0.0.6pre18.tgz is currently the right version to use? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a reliable T.38 Solution
On 01/05/2012 07:45 PM, Michael Keuter wrote: Am 05.01.2012 um 04:55 schrieb Matt Darnell: On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au wrote: I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and rx_fax on multiple installations with no problems. David, Are you running 10.0 or 1.8? Glad to know that the PAP2T has a solid T.38 implementation! -Matt There seem to be at least 2 versions of the PAP2T. The one I have (in Germany) does NOT support T.38. Michael http://www.mksolutions.info No PAP2 or PAP2T supports T.38, even though many people will swear that they do. For a little while there was some beta code for the PAP2T with badly broken T.38 support. Perhaps this is where the legend of T.38 on a PAP2T started. Of course, on the internet, when someone posts an incorrect message many people would like to believe is right, a 1000 people cite it as proven fact. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
On 10/08/2011 02:50 AM, Kevin P. Fleming wrote: On 10/07/2011 07:46 AM, Administrator TOOTAI wrote: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? You included the 'c' option to ReceiveFAX, telling it to act as the 'caller', even though it isn't the caller. This argument is parsed by ReceiveFAX in spite of it not being supported because the older app_fax version did support it, and we didn't want dialplans that included it to silently ignore the 'c' option. The same is true for the 'a' option; you'll note that neither of them are included in the documentation for the ReceiveFAX and SendFAX applications, and shouldn't be used. Why did you specify the 'c' option? Why was the ability to poll dropped from ReceiveFAX? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/08/2011 04:04 AM, Kevin P. Fleming wrote: On 10/07/2011 02:20 PM, James Sharp wrote: On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 Thanks for that; it helps. First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. t38UDPFEC is perfectly valid for version 0 of T.38. It works badly, so it makes no sense to use it, but it is valid. However, it is unlikely those are causing the call failure here. It's hard to say for sure without seeing the contents of the UDPTL packets, but based on their sizes, they are very likely t38-nosignal packets, and if that's all the FAX stack in Asterisk ever received, it would not trigger a FAX transaction to begin. Another possible problem is the repeated 'seq 0' in all the UDPTL packets, but this could be a problem with the UDPTL stack debugging messages themselves (this was just fixed in the Subversion branches for Asterisk 1.8 and later a couple of days ago). If you would, please retry this with the HEAD of the Asterisk 10 branch instead of 10.0.0-beta1, and also capture the UDPTL packets themselves so we can see what they contained. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
On 10/09/2011 02:38 AM, Ryan Wagoner wrote: On Sat, Oct 8, 2011 at 10:41 AM, Luke Hamburgl...@solvent-llc.com wrote: Interesting. I just signed up with Gafachi (haven't even tested the service yet) but I planned to make use of their T38 support since they are listed at voip-info as being one of the ITSP's that _do_ support T38. Have you tried contacting Gafachi with these results about their broken implementation? I would hope/expect them to try to fix this, instead of trying to force Asterisk to violate RFCs. It sounds like that Gafachi's T38 implementation is horribly, horribly broken I'm not tied to them at all, so if their stuff is broken, I'll go somewhere else. I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode and have tested with a Linksys SPA2102 and Zoiper. Gafachi basically told me they have many customers utilizing their T38 implementation and that it works. When asked for a list of compatible devices they said there were too many combinations and it was up to me to find a working solution. I am still looking a PAYG service provider that has a working T38 implementation. It seems like these are impossible to find. Ryan Gafachi was one of the few service providers to support T.38 when we first started providing T.38 support in Asterisk and Callweaver. We did get things working reliably with them, by making our software tolerant of a few weird things Gafachi do. Any practical T.38 has to be made to tolerate a lot of weird things other implementations do. So Gafachi has worked in the past, but its entirely possible they have now broken their service further. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
On 09/26/2011 01:01 AM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. You must be new here. It is Digium's long term hostility to reasoned input that means very few of the early contributors to Asterisk still contribute today. Steve Антон, Thanks. I will explore the option. -Bruce On Sun, Sep 25, 2011 at 12:05 PM, Paul Belanger pabelan...@digium.com mailto:pabelan...@digium.com wrote: On 11-09-25 01:54 AM, Антон Квашёнкин wrote: Just use cli aliases, provided by res_clialiases.so. 2011/9/25 Bruce Bbruceb...@gmail.com mailto:bruceb...@gmail.com Please don't feed the trolls. Thanks. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxes suddenly failing
Hi Tim, On 09/01/2011 03:49 AM, Tim King wrote: I realize that faxing is not great with voip but here is my confusion. I have been working on a web based fax system for 2 weeks. During this time I have sent over 100 2 page faxes without any errors. Now today as things are finally completed I can not seem to get any fax to go through unless it is a 1 page cover only. Anyone able to tell the issue from this debug output? -- Channel 'SIP/MyVoipProvider-0046' FAX session '12' started -- FAX handle 0: [ 000.38 ], STAT_EVT_STRT_RX st: IDLE rt: IDLENSRX -- FAX handle 0: [ 000.000184 ], STAT_EVT_RX_HW_RDY st: WT_RX_HW_RDY rt: RRDYNHRY -- FAX handle 0: [ 000.000504 ], P30EVN_RECEIVE_STARTED -- FAX handle 0: [ 000.000538 ], STAT_INFO_CSI -- FAX handle 0: [ 000.000568 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.091837 ], stack sent 5 frames (100 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.160248 ], stack sent 3 frames (60 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.960201 ], channel sent 48 frames (960 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 000.979464 ], channel sent 1 frames (20 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.157848 ], stack sent 150 frames (3000 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 003.219814 ], stack sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 005.240927 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 005.579811 ], stack sent 118 frames (2360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 006.481179 ], channel sent 275 frames (5500 ms) of silence. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 007.801045 ], channel sent 66 frames (1320 ms) of energy. -- FAX handle 0: [ 007.800554 ], STAT_FRM_CRP -- FAX handle 0: [ 007.800586 ], STAT_EVT_CRP st: WT_DIS_RSP rt: NT4X -- FAX handle 0: [ 007.800602 ], STAT_EVT_FSC_ERR st: WT_DIS_RSP rt: UNEXPECT -- FAX handle 0: [ 011.012832 ], STAT_EVT_RX_TRN_ENDst: WT_DIS_RSP rt: RXXXNFRX -- FAX handle 0: [ 011.012878 ], STAT_INFO_CSI -- FAX handle 0: [ 011.012905 ], STAT_INFO_DIS Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 011.152812 ], stack sent 279 frames (5580 ms) of silence. -- FAX handle 0: [ 013.179561 ], STAT_EVT_TX_V21_DONE st: WT_DIS_RSP rt: WDSRNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 013.471827 ], stack sent 116 frames (2320 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 014.260642 ], channel sent 323 frames (6460 ms) of silence. -- FAX handle 0: [ 016.119786 ], STAT_INFO_TSI Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.460661 ], channel sent 110 frames (2200 ms) of energy. -- FAX handle 0: [ 016.460394 ], STAT_INFO_DCS -- FAX handle 0: [ 016.460431 ], STAT_EVT_DCS st: WT_DIS_RSP rt: WDSRNDCS -- FAX handle 0: [ 016.460449 ], STAT_NEG_V17_14400 -- FAX handle 0: [ 016.460464 ], STAT_NEG_MH -- FAX handle 0: [ 016.460476 ], STAT_NEG_A4 -- FAX handle 0: [ 016.460488 ], STAT_NEG_RES_204x196 -- FAX handle 0: [ 016.460500 ], STAT_NEG_ECM -- FAX handle 0: [ 016.460514 ], STAT_EVT_SW_ECMst: WT_DIS_RSP rt: WDSRNSWE Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 016.540315 ], channel sent 4 frames (80 ms) of silence. -- FAX handle 0: [ 016.800906 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_TRN rt: UNEXPECT Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.700543 ], channel sent 158 frames (3160 ms) of energy. -- FAX handle 0: [ 019.759984 ], STAT_EVT_RX_TRN_ENDst: RCV_ECM_TRN rt: RTCFNERT -- FAX handle 0: [ 019.760071 ], STAT_FRM_CFR Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 019.912812 ], stack sent 322 frames (6440 ms) of silence. -- FAX handle 0: [ 020.957834 ], STAT_EVT_TX_V21_DONE st: RCV_ECM_STRT rt: RECMNT21 Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 021.278809 ], stack sent 68 frames (1360 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 022.261160 ], channel sent 128 frames (2560 ms) of silence. -- FAX handle 0: [ 022.517880 ], STAT_EVT_RX_IMG_STRT st: RCV_ECM_STRT rt: RECMNSRI -- FAX handle 0: [ 022.517982 ], P30EVN_PHASE_C -- FAX handle 0: [ 022.517998 ], P30EVN_DOC_START -- FAX handle 0: [ 022.518429 ], P30EVN_PAGE_START Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.102000 ], channel sent 442 frames (8840 ms) of energy. Channel 'SIP/MyVoipProvider-0046' fax session '12', [ 031.160415 ], channel sent 3 frames (60 ms) of silence. -- FAX handle 0: [ 031.160196 ], STAT_EVT_RX_IMG_ENDst: RCV_ECM
Re: [asterisk-users] Faxes suddenly failing
On 09/01/2011 11:50 PM, Lee Howard wrote: kirsten du toit wrote: You should try disabling ecm.. This seems crazy to me. Why are you recommending it? Because its the industry standard last resort of anyone who doesn't understand FAX and is using T.38. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 passthru on 1.8.5
On 08/31/2011 01:15 AM, Fabian Borot wrote: will installing spandsp help with t.38 pass-through? The only part of spandsp which is relevant to T.38 passthrough is its modem tone detection module, and I don't think the standard Asterisk distribution can make use of that. Some people do use it, to overcome the limitations in Asterisk's own tone detection, but I don't think they make their patches available. Steve From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 11:42:41 -0400 both endpoints use public Ips, I just changed the real ones for the privates ones to protect our ips but made a mistake and left the dest as a pub and the orig as private, my bad. but for the record, both are public IPs, there is no nat and iptables is off also, I see that the quintum sends a lot of these packages but asterisk sends only 1 or 2 to the other side. From: fbo...@hotmail.com To: asterisk-users@lists.digium.com Subject: T.38 passthru on 1.8.5 Date: Tue, 30 Aug 2011 09:44:15 -0400 Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 1.8.5.0 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway -- asterisk -- Dialogic IMG 1010 the call starts as a voice call, the remote fax picks up and we hear the fax tone, the we see the re-invite from the IMG asking for t.38, the RE-Invite is passed back to the user side [quintum gateway] whcih reply with 200 OK with t.38 and the nothing else happens. After 20 secs of inactivity the quintum sends another Invite with voice only and then a BYE. We do see that the quintum sends a lot of messages like this from the quintum's IP [192.168.1.18] but we do not see that asterisk sends the packages to the destination UDPTL (SIP/2345850624337933): packet to 192.30.189.146:12020 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) UDPTL (SIP/): packet from 192.168.1.18:10260 (type 0, seq 0, len 6) we have this settings on sip.conf faxdetect = yes t38pt_udptl = yes,maxdatagram=400 [I have tested with several combinations t38pt_udptl = yes;t38pt_udptl = yes,fec etc] When we send the fax from the quintum to the Dialogic IMG the fax works 100% of the times. I enabled fax set debug on and udptl set debug on but the console does not show almost anything but the udptl packets shown above. What else should I do?Any ideas/help is greatly appreciated txs a lot fborot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISAC and Asterisk
On 08/01/2011 07:43 PM, Kevin P. Fleming wrote: On 08/01/2011 04:12 AM, CB wrote: On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote: Are there any plans to include the ISAC codec in Asterisk? Is it possible or even desirable? Is ISAC open source (nothing indicates it is from the WebRTC website http://www.webrtc.org)? What do you need it for? The possibility of having a web-based softphone without requiring any plug-in is interesting. The adaptive nature of the ISAC codec could also prove useful. I see lots of possibilities in the mobile device space. I guess the lack of responses gives me the answer anyway! The IETF Opus codec is nearing completion, and it is very likely that it will be incorporated into the WebRTC stack soon after that. Given that, there's not much reason to spend time working on ISAC. A counter argument to that might be that Opus is fresh and new and nobody knows what patent issues might come crawling into view. iSAC has been around for a while. The source wasn't open until recently, but licenced users have had it for a long time. There has been much more opportunity for patent issues to show up. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ReceiveFAX app from FFA
On 06/21/2011 09:12 PM, khalid touati wrote: Ok, for the variables, I can retrieve some of them like the caller number and so on (I would assume that all the variables that last for duration of call are there), but I still think that I sould not use the h extension to continue after ReceiveFAX use, it's like not a lot of people use FFA, moreover very few came accross such an issue which is fine. Why do so may people think their problems are unique. Many people use FFA and spandsp. They all come across this. The issue is widely known, well understood, and not at all strange once you think about it. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with ReceiveFAX app from FFA
On 06/20/2011 03:38 AM, khalid touati wrote:
Hi Guys,
I solved temporarely my issue by kind of tricking Asterisk, I used the
following line instead of the old:
exten = h,n,System('/usr/local/
bin/fax2mail -p -f ${FAXFILENOEXT} --cid-number ${CALLERID(num)}
--cid-name ${CALLERID(name)} --dest-name Sir/Madam')
now when it hang up I receive my fax through email, and let me tell
you (first time using Free Fax from Asterisk) ReceiveFAX catch well
faxes, just a couple tries but got them all, let's see with more faxes
what will happen.
Why do you consider this a temporary fix? The far end machine will
normally hang up at the end of the FAX, so the hangup option in the
dialplan is exactly where you should expect to be.
If you need a couple of tries for some of your FAXes, it doesn't sound
like FFA is working very well for you. Check the timing of your
telephony channel. If you get more than 1% failures when sending FAXes
to and from your own equipment you should be looking into the cause.
Steve
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Re: [asterisk-users] OT - Which Android handset with Wifi-only ?
On 05/10/2011 12:55 AM, Olivier wrote: 2011/5/9 randulo rand...@randulo.com mailto:rand...@randulo.com On Mon, May 9, 2011 at 2:20 PM, mgra...@mstvp.com mailto:mgra...@mstvp.com wrote: Lots of Android handsets support wifi, like my G2, aka HTC DesireZ. Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not? Yes, of course, all dual-mode phones support WiFi but : 1. I'm not certain those would work without any SIM-card inside 2. those are likely to be more expensive than WiFi-only handset. See the last iPod touch which is marketed as a Sametime client is quite cheeper than the iPhone. To my knowledge, most Android-based WiFi-only machines are tablets. archos make cheap wifi only android devices, but I'm not sure of the small ones have the mic and speaker in the right place for a SIP call. There are some very cheap Android phones around, while the wifi VoIP phones tend to be expensive. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive faxes
On 05/06/2011 02:09 AM, David Backeberg wrote: T.38 has a boatload of problems, and most of those problems are because people who aren't employed by Digium did not read the specs, or they did read the specs, but felt like they had to violate the specs to get their code to work with a different broken T.38 stack. If you'd ever read T.38 you'd find what you've written there pretty funny. T.38 is full of holes. You simply cannot implement a working package from it. You have to experiment, find what other people have done, and try to fit in with that. The latest revision of T.38 is supposed to fill some of the holes, by incorporating text that Kevin Fleming, I and others prepared, as part of the SIP Forum working group that is trying to get the mess sorted out. However, due to the ITU's strange publishing procedures I have so far been unable to read this new revision. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive faxes
Unless someone has broken something recently, you'll get better results
with spandsp than you get with the Digium FAX package.
Steve
On 05/04/2011 09:21 PM, Satish Patel wrote:
Did you try digim fax ?
Also you can record you incoming fax via mxmonitor and analize it.
--
Sent from my iPhone
On May 4, 2011, at 8:50 AM, vip killa vipki...@gmail.com
mailto:vipki...@gmail.com wrote:
I've given up on trying T38 because there is no universal support for
it... Can someone recommend another way of faxing without using T38?
On Tue, May 3, 2011 at 5:13 PM, satish patel satish...@hotmail.com
mailto:satish...@hotmail.com wrote:
Enable debug and verbose on CLI ?
Did you enable and also at logger.conf
full = notice,warning,error,debug,verbose,dtmf,fax
Date: Tue, 3 May 2011 16:12:06 -0400
From: vipki...@gmail.com mailto:vipki...@gmail.com
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have full log.. only thing that stands out are two warnings:
[May 3 16:10:40] WARNING[18176] app_fax.c: Error transmitting
fax. result=13: Unexpected message received.
[May 3 16:10:40] WARNING[18176] app_fax.c: Transmission failed
On Tue, May 3, 2011 at 4:05 PM, satish patel
satish...@hotmail.com mailto:satish...@hotmail.com wrote:
I'd enable full debug at logger.conf and try to find issue.
-S
Date: Tue, 3 May 2011 15:55:51 -0400
From: vipki...@gmail.com mailto:vipki...@gmail.com
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
I tried with those settings and without... same error:
WARNING[18090]: app_fax.c:820 transmit: Transmission failed
On Tue, May 3, 2011 at 3:32 PM, satish patel
satish...@hotmail.com mailto:satish...@hotmail.com wrote:
did you set faxdetect=both or incoming
and faxbuffer=?
-S
Date: Tue, 3 May 2011 15:28:36 -0400
From: vipki...@gmail.com mailto:vipki...@gmail.com
To: asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] receive faxes
i have spandsp and app_fax.so is loaded but i get:
app_fax.c:820 transmit: Transmission failed
when trying to fax from a POTS line...
On Tue, May 3, 2011 at 3:27 PM, satish patel
satish...@hotmail.com mailto:satish...@hotmail.com wrote:
You need spandsp i guess following is my dialplan is
working example
[fax]
exten =
9000,1,Set(FAXFILE=/var/spool/asterisk/fax/fax.tif)
exten =
9000,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 9000,n,ReceiveFax(${FAXFILE})
exten = 9000,n,Hangup()
--
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Re: [asterisk-users] receive faxes
On 05/05/2011 03:29 AM, Lee Howard wrote: David Backeberg wrote: On Wed, May 4, 2011 at 12:00 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: (For my part, I'm actually surprised that nobody came up with a proper protocol for encapsulating the stream of zeros and ones that make up a fax transmission but rely on the precise timing inherent with a circuit-switched network, into something more suitable for sending over a packet-switched network. That would have fixed it good and proper.) They did. It's called TCP / IP. It allows sending PDFs, and they can even be encrypted. Faxing is for people who haven't heard of the internet. Nobody has said that faxing couldn't use TCP/IP... and there's no reason why T.38 couldn't use TCP/IP. Nobody has said that faxing couldn't use HTTP as a transport... or SSL... or any other kind of sensible mechanism. Why in the world people try to keep faxing (data transfer) tied-down to audio channels by putting T.38 into H.323 or UDP/IP SIP beats me. T.38 is defined to work over TCP/IP (although not TLS for some reason), but its rarely used. It can only really work between 2 T.38 boxes directly connected to the data network. To interwork with analogue FAX machines you need to maintain fairly tight timing, and that means sticking with UDP, as it does with all the other streaming stuff we do over UDP. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] receive faxes
On 05/05/2011 01:07 AM, Tzafrir Cohen wrote: Un-top-posting, On Wed, May 04, 2011 at 10:01:37AM -0400, vip killa wrote: On Wed, May 4, 2011 at 9:52 AM, Danny Nicholasda...@debsinc.com wrote: *You are “Running before you learn to walk”! You can’t make T.38 work (that’s ok, most other folks can’t either) but you want a free faxing solution that does multiple channels. Get the Free license and make that work, then pay Digium the $10 (or whatever it is) for the ports you think you need once the darn thing works.* screw that i just got hylafax to work with IAXMODEM...i refuse to pay digium a dime... supposed to be open-source right? Asterisk's fax support has two backends. One of them is FFA mentioned above. The other uses Steve Underwood's SpanDSP library and is completely free (speech, beer, whatever). You don't want to pay from the proprietary one, use the free one. Naturally those cheap bastards at Digium wanted so badly that you buy their FFA that they didn't bother writing the SpanDSP backend. Hmm... well, it seems they actually did. Well, in that case they surely don't include it in the binary packages they produce. Hmmm... they actually do. I've seen indications, such as at http://nerdvittles.com/?p=738 , that the spandsp support may not be working well these days. Can anyone comment on that, because all the bad stuff I've seen on this mailing list about FAX in 1.8 is breakage of FAX detect and the Digium FAX module? That is not to say IAXMODEM is not a cool project on its own. Certainly HylaFax+IAXModem is the right tool for certain scenarios, and a useful tool generally. Cheers, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Export Fax from Wave file
On 04/21/2011 08:12 PM, Khaled W. Chehab wrote: Dears, I configured an account on my asterisk pbx to record the outgoing calls. When the asterisk pbx user make a call and send a fax the call recorded to wave file format. I searched the internet and found a software that can play the recorded wave file and export from it the tiff fax document sent. Is there a way that asterisk can play the wav file and export the tiff document ??? If you have found software to do this, what are you looking for now? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8.4-rc2: ReceiveFAX fails
On 04/16/2011 08:47 PM, Ryan Wagoner wrote: On Sat, Apr 16, 2011 at 1:56 AM, Steve Underwoodste...@coppice.org wrote: On 04/16/2011 07:25 AM, Ryan Wagoner wrote: On Fri, Apr 15, 2011 at 7:00 PM, sean darcyseandar...@gmail.comwrote: Using spandsp-0.0.6-pre18, the Jan 22 release. You might try using spandsp-0.0.6-pre17. That version works great for me with 1.8.4-rc2. When I tried pre18 it failed to receive any faxes. Of course, such an important regression was duly reported to the author, wasn't it. Steve I wasn't sure if it was my problem or a regression with the release. When I had searched nobody else mentioned the issue. I'd guess I have around a 70% success rate with pre17 over ulaw from free fax services. When I briefly tested pre18 I couldn't get any to come through. I do appreciate your effort with spandsp. 70% is awful. You should be getting 99%, unless this is VoIP over a sucky network. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sox and bad quality when converting to 8 kHz
On 03/25/2011 04:58 AM, Thomas Winter wrote: Hi list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help best regards Thomas You really need to remove the bass end of the spectrum before downsampling to 8k/s. uLaw/ALaw sound pretty muddy and horrible if you don't do that, and the other common 8k/s codecs don't sound any better. Jean-Marc Valin wrote a little filtering utility for this purpose. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.711.0
Hi, Has anyone seen G.711.0 in real world use? The spec was published quite a while ago, but as far as I can tell there is no RFC defining the SDP and RTP details needed to deploy it, and nobody advertises that they support it in their products. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using voice modem as poor man's FXO in Asterisk 1.8
On 02/28/2011 10:12 AM, Stuart Longland wrote: Hi all, I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. The web server in question is an Intel Atom system with a Mini-ITX motherboard. Its one and only PCI slot is occupied by a PCI ethernet card. So FXO card is not an option even if it were within budget. My options therefore look to be an external FXO device of some description (Ethernet or USB), or to use a voice modem. I fear external FXOs are going to be even more expensive than internal FXO cards. Now, I have here an old Maestro JetStream 56k modem here that does amongst other things, voice comms, and I have used it in the past as a telephone by plugging a headset into the front of it (and it was full duplex too if I recall correctly). I have also used it as an answering machine, with the audio being transmitted digitally over the RS232 link. So that to me suggests it is possible to get audio in to and out of the modem, either via a sound card or using the serial port. The web server has a sound card too (hard not to buy a motherboard with one these days). Apart from the lack of any hardware signal processing, it seems all the components are there. The server isn't particularly heavily loaded, and thus I see no reason why the machine wouldn't theoretically be able to handle the DSP in software … I've seen lesser hardware do quite sophisticated DSP in real-time. Now, I've hunted high and low for where this is configured. Some mailing list threads point me to the nonexistant /etc/asterisk/modems.conf. One points me to /etc/asterisk/phone.conf, but nothing there jumps out at me as being an obvious means for configuring a modem — nor can I find where it's documented on the Asterisk wiki. Where abouts should I look for documentation on configuring these modules? Regards, There is no requirement for DSP. There is a requirement for getting duplex audio in and out of the PC. *Very* few full blown external modems will do that. The very simple USB winmodems will, but nobody has produced drivers to make it work for any of the common chips used in those devices. Its not hard to do, though. Source code exists which is not a million miles from that required to hook a USB winmodem into DAHDI. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any voice changer applications for Asterisk?
On 02/06/2011 05:05 PM, Sherwood McGowan wrote: AAhem. https://wiki.asterisk.org/wiki/display/AST/Function_PITCH_SHIFT Granted, it's in 1.8, but it's in the documentation ;-) Cheers That seems to do exactly what the Lobstertech code does. What do people use this for? The Lobstertech one was a fun toy, but seems to be of no practical use. Changing female to male, child to adult, etc. seems pretty useful, but these modules make no attempt to perform a meaningful voice change. They would need to control the formants independent of the pitch to produce anything like a plausible voice adjustment. On Sat, Feb 5, 2011 at 9:44 PM, Steve Underwood ste...@coppice.org mailto:ste...@coppice.org wrote: On 02/06/2011 05:39 AM, Bruce B wrote: Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. It might help if you explained the kind of change you would like to make, which the lobstertech module doesn't offer. Steve Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any voice changer applications for Asterisk?
On 02/06/2011 05:39 AM, Bruce B wrote: Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. It might help if you explained the kind of change you would like to make, which the lobstertech module doesn't offer. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp download
On 01/22/2011 01:00 PM, Bryant Zimmerman wrote: Where can I get the latest stable version of spandsp. That will work with res_fax_spandsp.so. The link listed on the voip-info website is dead. Any other location for download? http://www.soft-switch.org/ There was a server failure. It should be back up now. Use 0.0.6pre18 Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
On 01/21/2011 08:37 PM, Tom Rymes wrote: On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote: A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. As a longtime Hylafax user, I can confirm it's an excellent solution. I am somewhat surprised about the comment of being able to do audio or t.38, but not both. This is probably a little true and untrue at the same time, though I have never used t.38modem with Hylafax. Given the structure of the product, you could have HylaFAX connected to both an IAXModem and a T.38Modem at the same time (or 23 IAXModems, a 24-port T1/E1 PCI-card modem, and 7 t.38modems for that matter...). What it cannot do, is receive audio and t.38 on the same port, which is what I presume that Steve was referring to. This is really a limitation of IAXmodem and t.38modem, as one only handles audio, the other only handles t.38. In other words, you could route t.38 faxes to it on port 1 and audio faxes on port2, but you cannot have port 1 handle both types. Its easy to set up some t38modem channels and some iaxmodem channels for receiving FAXes. Transmit is more problematic. With this split config, you need to know in advance whether the particular number is accessible by T.38 or by audio. Most people won't. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ReceiveFax
On 01/20/2011 11:00 PM, Flavio Miranda wrote: Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Receivefax can handle hundreds of calls at one time, if your machine's resources are up to it? Why would there be a restriction of one call? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax
On 01/21/2011 06:46 AM, Bryant Zimmerman wrote: On 01/20/2011 11:47 AM, Steve Underwood On 01/20/2011 11:11 PM, Kevin P. Fleming wrote: On 01/19/2011 02:30 PM, Bryant Zimmerman wrote: On 01/19/2011 02:05 PM, Bryant Zimmerman wrote: I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. *From*: Jason Parker jpar...@digium.com *Sent*: Wednesday, January 19, 2011 3:19 PM There was a typo in the res_fax documentation. Application_SendeFax should be the correct documentation. I don't know where Application_SendFax is coming from - it's probably old. When the next import happens, Application_SendFax should be replaced by the correct version (then I'll try to remember to remove the bogus SendeFax copy). Jason thanks for the clarification on this. If I start my development with the res_fax_spandsp.so module. Should all of my code be compatible with the res_fax_digium.so module? I want to be able to get things running and tested and move to the digium supported option in the future. The choice of technology module is mostly irrelevant; that was the whole point of splitting res_fax out from them. If you use the applications and other features of res_fax, it won't matter which underlying technology module is loaded. Well, people do get problems with the Digum FAX software, which go away when they switch to spandsp. Its best to test with the code you intend to deploy. Steve Steve is there any real compelling reason to res_fax_digium.so over the res_fax_spandsp.so? I was thinking Digium module was likely to be better is this wrong based on what people are seeing? Feature wise they are similar, using an Asterisk release. By adding patches from the bug tracker, spandsp can work as a T.38 gateway, which the current Digium code cannot. I assumed by now Digium would have launched a V.34 version of their FAX module, which is something a free version can't do for a few more years, but there seems no sign of that happening. People tell me spandsp is more flexible in its TIFF file handling, but I've never found any documentation on what the Digium file handling is supposed to be capable of. Speed wise I have no comparisons. There are people running hundreds of concurrent FAXes all day using spandsp on quad core servers with good disk setups. I have no idea how fast the Digium software can be. Performance wise I've helped people get off the Digium FAX software, and start using spandsp, to get around problems. A couple of people were frequently finding only the first 1/4 or so of each page in the output file, when the received T.38 stream was perfect (i.e. I could play a PCAP of the session into spandsp, and get a perfect TIFF file). Those people complained that the only support offered by Digium was an offer of a refund. I've help a couple of people who regularly see weird T.38, which the Digium FAX was handling in a very ungraceful way. Spandsp handled it badly too at that time, but the latest spandsp snapshots do a good job. To be fair, I only get contacted when the Digium FAX software screws up, Digium are no help, and the person is looking for a solution. I get little visibility when spandsp might do something bad, and the Digium software does a better job in the same situation. A comparison wouldn't be complete without mentioning Hylafax. Hylafax has a great infrastructure - tools for integrating with Windows clients, and so on. Neither spandsp or the Digium FAX code can match that for FAX termination. I think its biggest drawback is you either use it with iaxmodem for audio FAXing, or t38modem for T.38 FAXing. It can't smoothly integrate the two right now. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax_digium.so crashing
On 01/17/2011 04:37 AM, Jeremy Kister wrote: Since digium is apparently blind to users of their Free Fax for Asterisk, does anyone have advice on how to report a crashing problem with res_fax_digium and Asterisk 1.8.2 ? Use spandsp. I have detailed logs/reports and a backtrace ready, but I have no idea who can help. Nobody, if you don't post them somewhere. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/08/2011 03:44 AM, Kevin P. Fleming wrote: On 01/06/2011 11:34 AM, mgra...@mstvp.com wrote: We should also be very clear that the Siren codecs are supported on the Polycom SoundStation conference phones and the VVX-1500 Business Media Phones. These codecs are not supported in the SoundPoint desk phones. The SoundPoint series support the more basic G.722 codec in the IP335/450/550/560/650/670 models. The SoundPoint IP6000 and IP7000 conference phones (and maybe the IP5000, I haven't checked) also support G.722.1 and G.722.1C. The IP6000 is actually model Polycom recommended for testing when we implemented G.722.1. One of the annoying things about the Polycoms is trying to work out what they can do. You have to search quite hard to find which codecs each model supports. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 05:25 AM, Tim Panton wrote: On 5 Jan 2011, at 13:07, Steve Underwood wrote: G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering 14kHz bandwidth. These are most often found in Polycom phones, but they are available elsewhere. The only widely supported HD codec is G.722. Pretty much anything offering wideband voice supports G.722. Except skype which only supports SiLK as the HD codec. I mention this because most people's experience with HD will be in a Skype-to-skype call, although admittedly not in this group. That's a very good point, although Skype does support more codecs than just Silk, and I believe G.722 may be one of them. Nonetheless, it is Silk that people have got used to. It offers about 11kHz bandwidth, so it is wider band than G.722. The critical addition than wideband gives over normal telephony is the 5kHz to 7kHz area, where a lot of the energy that allows us to differentiate the unvoiced phoneme lies. The energy between 7kHz to 15kHz does, however, add a lot to the human voice, and allows for a more relaxed listening experience - its just less tiring to listen to. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/05/2011 03:29 PM, Bruce B wrote: 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)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. G.722.1 is a 7kHz bandwidth codec. G.722.1C is a stretched version offering 14kHz bandwidth. These are most often found in Polycom phones, but they are available elsewhere. The only widely supported HD codec is G.722. Pretty much anything offering wideband voice supports G.722. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 01:04 AM, Tilghman Lesher wrote: On Wednesday 05 January 2011 07:07:10 Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: 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)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. The Siren7 and Siren14 codecs in Asterisk are licensed code from Polycom, so they are indeed the Siren7 and Siren14 codecs. They will interoperate with any other vendor who has licensed those codecs from Polycom. What Polycom licence to everyone is actually G.722.1 and G.722.1C. Been there. Done that. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Are the Siren7 and Siren14 the G.722 HD voice codecs?
On 01/06/2011 12:05 AM, Kevin P. Fleming wrote: On 01/05/2011 07:07 AM, Steve Underwood wrote: On 01/05/2011 03:29 PM, Bruce B wrote: 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)? The G.722 codec in * is G.722. The Siren7 codec in * is probably not Siren 7, but G.722.1. G.722.1 is very similar to Siren7, but uses a different code in the SDP and has some minor differences in the codec. The name G.722.1 may look similar to G.722, but the codecs bear no relation to each other. The Siren14 codec in * is probably not Siren14, but G.722.1C. G.722.1C is very similar to Siren14, but like Siren7/G.722.1 the SDP code is different, and there are minor differences in the codec. Asterisk actually supports both the Siren* and G.722.1* names in SDP negotiations. I wasn't aware there were bitstream incompatibilities between the Siren* and G.722.1* variants, even though the code may be slightly different... so Asterisk uses a single codec module for both variants. I am unclear how compatible or incompatible the bitstreams may be. What I know (from implementing these codecs) is that the source code Polycom provide licencees, as the basis for developing their own G.722.1 and G.722.1C codecs, has several comments referring to things not being quite the same as Siren7/Siren14. However, they don't hand out the actual Siren7/Siren14 source code, so I don't know how much divergence there is. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/05/2011 02:39 AM, Tom Rymes wrote: On 01/04/2011 8:55 AM, Kevin P. Fleming wrote: On 01/03/2011 06:47 PM, Thomas Rymes wrote: On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: [snip] OK. Either way, though, the changes to echo cancellation are not affected by the faxdetect setting, right? That is correct; the faxdetect setting and the echo canceller behavior are completely unrelated. Excellent. [snip] Is there a time limit to when DAHDI listens for faxes (say the first 10 seconds of a call?), or might it detect one in the middle of a ten minute call? I haven't double-checked, but I believe the software DSP will be in place on the call until it sees a CNG tone, regardless of when that happens during the call. Wouldn't it make sense to be able to specify a time period after which chan_dahdi disables fax detection? Only calls that begin with a voice call and end with a fax would benefit from detection after the initial ~8 seconds of a call, unless I am overlooking something. If the DSP keeps listening and detects a spurious fax tone (I know I have seen the human voice incorrectly identified as CNG), it will send the call off to the fax extension if one exists in the same context. In fact, we ran into some issues with exactly that happening. It is very normal for many people to chat and then start their FAX machines, especially domestic FAX users with a FAX machine attached to their home land line. If you don't care about those your proposal is OK, otherwise. There is no excuse for false detection of FAX tone. It takes a very poor detector to mistake voice for FAX, unless the person is specifically trying to whistle the right tones (which some people are quite good at). [snip] Thanks for the clarification, there's a lot of conflicting info out there. Feel free to comment on wiki.asterisk.org if any of the information there led you astray; we'd like to get that to be the most accurate place for people to find this sort of information. I'll give it a look. I had not specifically looked at the asterisk wiki, but Google searches brought up lots of messages confusing the fax operation of the echo canceler with the faxdetect= setting for DAHDI/Zaptel. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/04/2011 09:53 PM, Kevin P. Fleming wrote: On 01/03/2011 07:08 PM, Steve Underwood wrote: On 01/04/2011 04:22 AM, Kevin P. Fleming wrote: No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. That's not true. Modern echo cancellers normally disable completely. It is arguable whether they should disable completely for FAX, but they need to behave properly for all modems. For any duplex modem, disabling only the NLP is useless. They need to cancel end to end, so they don't get upset by a continuously adapting canceller, and so they can minimise the issues caused by the highly non-linear G.711 channel. This doesn't match up with what the manufacturers of the two G.168 ECs that Digium distributes have told me personally about their products. Their ECs behave differently for FAX and 'regular' modems, but they do that based on the detection of the V.21 preamble, ANSam and other signals in addition to CED, which seemed to be much more detail than was warranted in my response to the OP :-) Well, that makes a bit more sense, but I am very skeptical about this. The Octasic canceller is highly problematic with various modems and tones, so they aren't exactly a reference model for how to do things. Reports I here of the other canceller are much more positive. Its obvious why they want to keep the canceller alive. Long echoes over VoIP channels, combined with slow responding FAX boxes, can lead to a FAX machine hearing its own output heavily delayed, and it may mistake this for the response from the far end. T.38 largely avoids this kind of issue. The start of a FAX call doesn't really have a good signal on which to train a canceller. They can use the first V.21 burst in each direction (The FAX signals for G3 or the V.8 exchange for Super G3), and then lock down the canceller, but those signals aren't wide band enough to be ideal. The canceller could adapt very oddly. If they continue adapting once the wideband signals from the fast modems start, they are likely to upset modem operation there. If they just accept that, and rely on the fast modem retrying, it will usually step down in speed. I believe I have seen this behaviour in setups where the signal looks very clean, but the FAXes always exchange at 12000bps. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: I'll try. 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. That's not true. Modern echo cancellers normally disable completely. It is arguable whether they should disable completely for FAX, but they need to behave properly for all modems. For any duplex modem, disabling only the NLP is useless. They need to cancel end to end, so they don't get upset by a continuously adapting canceller, and so they can minimise the issues caused by the highly non-linear G.711 channel. 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. If the CNG tone arrives from the network side of the DAHDI channel (the far endpoint), then yes. 3.) faxdetect=outgoing will ?? The same thing, but if the CNG tone is being sent towards the DAHDI channel (from the near endpoint). This is rarely used. Also, do Digium cards with HW Echo Cancellation detect the CNG tones in hardware? If so, how does the faxdetect setting in DAHDI affect that behavior? No, none of the Digium HW ECs detect and report CNG tones via the DSP; CNG tone detection is still done on the host CPU. 'faxdetect' is not set in DAHDI, it's set in chan_dahdi.conf. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729a and G729 interoperability
On 12/27/2010 08:05 PM, Elliot Murdock wrote: Hello! I am wondering how the differences between G729, G729a, and G729b effect call bridging and server interoperability. For example, can one server use the G729 code with another server that uses the G729A codec? Also, which version is Asterisk set up to use? Thanks! Elliot There is no compatibility issue between basic G.729 and G.729A. That is why they use the same SDP code. In practice it is rare to see a G.729 codec in real world use. They are almost all G.729A. G.729 sounds better, but G.729A uses half the CPU power. Cheaper normally wins over better in the real world. G.729 annex B is an add on, providing VAD features. It may be used with G.729 or G.729A. A separate entry in the SDP says whether this option is supported. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem
Hi Michael, Use spandsp. It is more relaxed about the file resolution, to avoid this exact issue. Files with a resolution within 5% of 204x196 are accepted. However, if you have really made the image width 1680 pixels, that is wrong and I would be surprised if any FAX software accepts it. Standard sized FAX images are 1780 pixels wide. Steve On 11/20/2010 06:02 PM, Michael wrote: Hi, We played around with the different parameters of the tif files and found that the issue was with the resolution. Most files generated on the PC have a 200x200 resolution, but it seems that FFA only accepts 204x196 resolution. Right now, we added a process to change the file resolution using ImageMagick, but it would make sense to allow also 200x200. Michael Original Message Subject: Re: [asterisk-users] FFA (Fax For Asterisk) tif file (size) problem From: Mark Deneenmden...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 19 November, 2010 17:19:48 On Fri, Nov 19, 2010 at 9:42 AM, Michaelvoip.quest...@gmail.com wrote: Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error FAX handle 0: failed to queue document 'filename.tif', so we set it to 1680x2285, but it's still rejected. Is there a way to debug this further and fix it? We often have tif source files that we prefer to send, without converting to pdf and back to tif. Thank you in advance, Michael I don't know if this is the case or not, but check for differences between the two tiff files. I wonder if one is compressed and the other is not? -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for a better ATA
On 10/09/2010 06:36 AM, Jeff LaCoursiere wrote: On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant I'm curious which of the above ills you attribute to the Linksys (assuming an SPA2102? The PAP2T does have the T38 problem I believe). Its basically the defacto standard for all the giant ITSPs. Perhaps your problem is one that could be rectified in some way. I have also tried Grandstream and Audiocodes (still use the MP-124s in certain situations) and have found that the SPA2102s work the best for us... The PAP2 and PAP2T do not support T.38. The SPA2102 and SPA3102 support it, but have a number of quirks. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3rd party app store
On 09/19/2010 12:06 AM, Darren Nickerson wrote: On Sep 18, 2010, at 11:41 AM, Mark Deneen wrote: On Fri, Sep 17, 2010 at 11:52 PM, Dean Collinsd...@cognation.net wrote: Any thoughts on why the lack of traffic? Cheers, Dean Not enough applications to play immature bathroom sounds. You could well be right, but consider for a moment a few alternatives. Perhaps it's the $5000 up front just to be listed? I see the fee's reduced to $2500 now as a promo, but still that's a huge barrier for most. Even $1 will keep most free solutions out of a forum like that, so a blanket fee strategy must have been specifically chosen to skew things in a particular way. Seems like it worked very well. Or perhaps its the fact that the nature of the apps that get listed means they aren't usually 'purchase-able' with a simple 'click to buy' (how do you sell SIP trunking with a click-to-buy???) - and as a consequence there's no purchase capability built into the asteriskexchange site, just link outs to different purchase-ish URLs for the various products. Anyone looking to sell their app would need to develop their own point-of-sale/payment processing systems so it's really not an 'app store' at all in the traditional sense. That is a pretty basic problem for some things, but not for everything. Plenty of telephony stuff is a thing for sale, even if some after sales support is needed, to get over installation issues. Kudos to digium for realizing this goal, but I think the $5000 get-in cost has resulted in the lack of interest/popularity, and limited the listings to only the largest, most profitable asterisk/digium partners. Kudos to Digium for taking an idea that could have worked against their interests, and sidelining it so well nobody created a real marketplace. The bottom line, of course, is that if people like regular posters here didn't know about about the site, the real target audience most certainly does not. Nothing more is needed to explain the low traffic. Even if you are serious about creating a vibrant, orderly marketplace, its really hard. Look at the variation in quality between them. Even Google, which is basically a marketing company, seem to have no idea how to make the Android market function. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 and fax
On 09/14/2010 04:33 AM, Stanislav Korsei wrote: Hello! I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5. When i try to receive fax I get: Why install 0.0.5? Its ancient. the world has moved on. [Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel 'SIP/crocus-ua-0004' refused to negotiate T.38 [Sep 13 00:46:02] WARNING[3283]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Sep 13 00:46:02] WARNING[3283]: app_fax.c:817 transmit: Transmission error I definitely know that this peer supports T.38 because it works on Lynksys PAP2T. The Linksys PAP2T does NOT support T.38, so this statement makes no sense. The Linksys SPA2102 and SPA3102 support T.38. The PAP2 and PAP2T do not. Dialplan is such: answer() wait(6) ReceiveFAX(/var/spool/asterisk/test.tif) Am I doing something wrong here? Apparently. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving from DSL to T1
On 09/14/2010 04:23 AM, Joel Maslak wrote: On Mon, Sep 13, 2010 at 1:12 PM, Hans Witvliet h...@a-domani.nl mailto:h...@a-domani.nl wrote: No these are also geo-stationary (same altitude, so same delay), commercial and military satelites, Yes, exactly. Geostationary satellites have been used for telephone for ages (and are still used for remote areas - they have advantages over the disintegrating constellations such as iridium - namely predictability). When geostationary satellites were the normal thing for intercontinental calls, the call was normally satellite one way and cable the other. Satellite both ways would have been cheaper, but the total round trip latency was go bad, it was hard to hold a proper conversation. As for consumer (home) grade satellite internet service, it's pretty low quality. But if you have money, you can have just as good of service as the telcos enjoy for TDM voice over them (even with VoIP). I know several organizations using them (but they are paying more than the $100 or so a month as is typical for a home user - a lot more). Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible malformed G729B - SID (VAD/DTX)framesfrom carrier endpoint ?
On 09/07/2010 12:03 AM, Jeff Brower wrote: Steve- We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. We have Kamailio + rtpproxy running between the endpoints. Do you think it's reasonable to convert the first malformed SID frame (10 bytes) to 2 bytes, and then strip the following malformed SID frames until we see the talkspurt marker bit is set? We could do that... I'm wondering if anyone has seen such malformed SID frames before. As a couple of additional notes, between us and the remote endpoint there appears to be using an ALOE Systems (formerly MERA systems) MSiP system. So far the SDP negotiations we've tried (e.g. a=fmtp:18 annexb=no) have not convinced the remote endpoint to disable VAD. What makes you think there is a SID with the wrong length, rather than no SID? Do the first 2 of the 10 bytes look like SID? The first two bytes appear not to be a proper SID. However, as Vikram mentioned time-stamps show an increase greater than ptime and MARK bit is set in the RTP header. Then there are several consecutive packets (from 10 to 100) with this combination. Once we see the first of these, possibly we could strip and generate a correct SID. You can't generate a correct SID. The codec constructs the SID information from its working variables, and may send extra SID messages during the silence period, to update the model it sent in the original SID. I expect if you have annexb set to no, then some other form of VAD is active, and suppressing transmission. Yes... something in the middle... possibly the MSiP. -Jeff Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What can make G.729a codec hostid change?
On 09/08/2010 03:23 AM, Gordon Henderson wrote: On Tue, 7 Sep 2010, Tiago Geada wrote: Hi, I don't have any g729 codec license. But by reading Barry's complaint I get to think that it is really unfair that Digium can't renew his license or something. I am a Debian user myself and I understand the need to upgrade from etch to lenny (and to squeeze in no time). Having a kernel built on purpose to remove some modules is out of line. A better solution needs to be provided in cases like these. Buy licenses from Digium, get the software from Latvia... Gordon If you do that your Latvian software is unlicenced. The patent licencing terms for G.729 mean if you buy from someone (e.g. Digium) you are only licenced to run the implementation (software or silicon) they provided. If you want to licence some other code which is currently provided without patent licencing, you need to contact the licencing pool, and meet their conditions. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible malformed G729B - SID (VAD/DTX)frames from carrier endpoint ?
On 09/06/2010 11:18 PM, Jeff Brower wrote: Steve- On 09/05/2010 04:08 AM, Vikram Ragukumar wrote: Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. We have Kamailio + rtpproxy running between the endpoints. Do you think it's reasonable to convert the first malformed SID frame (10 bytes) to 2 bytes, and then strip the following malformed SID frames until we see the talkspurt marker bit is set? We could do that... I'm wondering if anyone has seen such malformed SID frames before. As a couple of additional notes, between us and the remote endpoint there appears to be using an ALOE Systems (formerly MERA systems) MSiP system. So far the SDP negotiations we've tried (e.g. a=fmtp:18 annexb=no) have not convinced the remote endpoint to disable VAD. What makes you think there is a SID with the wrong length, rather than no SID? Do the first 2 of the 10 bytes look like SID? I expect if you have annexb set to no, then some other form of VAD is active, and suppressing transmission. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?
On 09/05/2010 04:08 AM, Vikram Ragukumar wrote: Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP packets with encoded G729 payload. VAD/DTX is enabled. We see that the last frame transmitted by the carrier side endpoint, before the beginning of a period of discontinuous transmission has 20 bytes of payload. We have verified that VAD/DTX is used by the carrier side endpoint by noting that there exist successive RTP packets that differ by 1 in their sequence number but have a timestamp difference 160 and MARK bits are set in the RTP header. Our understanding is that for G729B, the SID frame that is transmitted before a period of discontinuous transmission has a size of 2 bytes. However we see that ALL RTP packets sent by the carrier side end point has a length of 20 bytes. Has anybody else seen this behavior from a carrier side endpoint ? Is there an RFC or document that specifies Your understanding is correct. You need to infer from the length of the last frame being 2 bytes that it is a SID frame, and SID frames should only ever occur as the last frame in an RTP packet. If the SDP negotiation has agreed to used the annex B (CNG/DTX/VAD) option for G.729 you would normally expect to see a SID frame at the end of transmission. If the SDP negotiation has agree to do CNG/DTX/VAD by another means (which it can do) you won't see those SID frames. Even when annex B is used, I think some systems may miss out the SID frames. The use of proper annex B processing requires additional patent licence payments, and I suspect some people try to fudge things to save a little cost. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk compatible cards?
On 08/10/2010 09:40 PM, Jeremy Betts wrote: I have always had very bad experiences with the x100p cards, they always have very bad echo. If you need decent call quality I would wait until you can afford a Digium card. Use OSLEC with them, and they work OK. Even if they don't have a selectable impedance to match local conditions. They certainly work a lot better than things like the SPA3102, which has very screwed up echo handling. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk compatible cards?
On 08/10/2010 11:18 PM, Seann wrote: Steve Underwood wrote: On 08/10/2010 09:40 PM, Jeremy Betts wrote: I have always had very bad experiences with the x100p cards, they always have very bad echo. If you need decent call quality I would wait until you can afford a Digium card. Use OSLEC with them, and they work OK. Even if they don't have a selectable impedance to match local conditions. They certainly work a lot better than things like the SPA3102, which has very screwed up echo handling. Steve In everything I have read you don't use Echo handling on the SPA3102. I own one and haven't had a problem with it and Asterisk, but I use Asterisk to handle any echo on a software level, and don't handle it with the SPA3102. ~Seann Many people suggest you turn off echo cancellation on the FXO port of a SPA3102 because its so badly broken its worse than nothing. Most people do need cancellation on that port, though, and have nasty echo problems. Just Google for echo and SPA3102. Some ITSPs have ripped them out and dumped them because they are so much trouble. There is nothing you can do within Asterisk about echo on an SPA3102's FXO port. I have no idea what you mean by Asterisk handling the echo on a software level. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
On 08/06/2010 04:43 PM, Jeff Brower wrote: Steve- On 08/06/2010 05:40 AM, Jeff Brower wrote: Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. I think all the IP for MELP is now in the hands of Compandent, and TI no longer has the ability to waive royalties. Either way, government use and use with TI silicon are two niches that might work out well, and everything else is a problem for several more years. If you are going to pay royalties for a low bit rate codec, IMBE is probably a better option. TI is a good option, but what do you have against Mindspeed? Choosing a good option for this kind of card is mostly about managing the patent licence fees. I assume Mindspeed gave Digium the best option for doing that, within Digium's volume constraints. so there is still a place for LPC10 [...] I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. MELPe is definitely a compandent thing, and TI cannot waive fees for that. MELP and MELPe are derived from LPC10. Any attempt to improve LPC10 would take you down a similar road, though you would need to skirt around the patents. Do you really consider MELPe to be an enormous improvement over LPC10? Its still pretty lousy compared to a number of options at about 5kbps, and RTP overheads mean the gain from going lower than 5k isn't that big. The main reason LPC10 and MELPe offer a low bit rate in RTP is the minimum packet you can pack 22.5ms frames into sanely is a 90ms one. 90ms RTP *really* cuts the overheads, compared to the more typical 20ms or 30ms packets used for G.729. As others have mentioned, David Rowe is working on a modern 2400bps codec. He did a burst of work some time ago, and then put it aside while busy with other things. He recently told me he is restarting the work, and he wants to get that codec into good shape before the end of this year. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Conversion
On 08/07/2010 03:15 AM, Jeff Brower wrote: Steve- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.comwrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, Not if used for govt/defense purposes. For commercial-only purposes, TI will waive royalty fees if their chip is used in the product. It would have been nice if Digium had considered the many advantages of using a DSP pioneer such as TI before putting a Mindspeed chip on their TC400B card. I think all the IP for MELP is now in the hands of Compandent, and TI no longer has the ability to waive royalties. That is not correct. Compandent has filed copyrights on certain files associated with a C549 chip assembly language implementation they did under contract to NSA around 2001. TI has patent rights on 2400 bps, TI + Microsoft on 1200 bps, and TI + Microsoft + Thales Group on 600 bps. Microsoft's IP came about as a result of acquiring a company called SignalCom around 2001. If the noise pre-processor is used, then there is some ATT IP. To verify this, you can search dsprelated.com (specifically, look for posts discussing this issue on comp.dsp), and you can also read the Compandent IPR section of the MELPe Wikipedia page (http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction). That section was authored by the Compandent's founder, Oded Gottesman. Oded is a super sharp, very hard working guy. Compandent also claims a copyright on some C code in the file melp_syn.c (synthesis filter). I have read discussions by DSP experts indicating the copyrighted section of code can be implemented in alternative ways, but Oded may say that's not accurate. That guy is PITA. He must have driven a lot of people away from MELP by the way he acts. He really annoys the regulars in the comp.dsp group by posting astroturf questions about MELP, and giving astroturf replies about how fantastic it is. That probably shapes a lot of my attitude to MELP. :-) Either way, government use and use with TI silicon are two niches that might work out well, and everything else is a problem for several more years. If you are going to pay royalties for a low bit rate codec, IMBE is probably a better option. I would disagree because IMBE source is not available. MELPe source is available and can be downloaded online. Depends what you mean by available. IMBE is patented, just like MELP is patented. Licence either, and implementations are available. IMBE has the great benefit of being widely used for commercial and amateur low bit rate channels. For example, amateur radio uses IMBE - an anomaly which is one of the drivers for David Rowe's work on an open low bit rate codec. Transcoding at low bit rates is a disaster, so using a codec you won't need to transcode is a big plus. TI is a good option, but what do you have against Mindspeed? Choosing a good option for this kind of card is mostly about managing the patent licence fees. I assume Mindspeed gave Digium the best option for doing that, within Digium's volume constraints. My understanding in talking to Digium engineers at Globalcom and other trade shows back in 2006 is they were worried about interfacing the TI TNET series devices over the PCI bus. They would have needed an FPGA with some non-trivial logic programming, so I understand their decision. But if they had got past their FPGA writer's block, they could have put one TNETV3010 chip on there, even smaller than the Mindspeed and without the heat sink, and had twice the channel capacity as they do now. TI have had DSP chips with a PCI interface for years, so that explanation doesn't make a lot of sense. Of course, these days you need a PCI-E interface. I'm not so sure about the status of those in DSP chips. so there is still a place for LPC10 [...] e I haven't seen an LPC10 implementation with MOS higher than 2.5. Due to its age and expiration of patents, LPC10 might be a basis for a 2400 bps open source codec. But enormous improvement would be needed to come close to MELPe performance. MELPe is definitely a compandent thing, and TI cannot waive
Re: [asterisk-users] Codec Conversion
On 08/06/2010 05:40 AM, Jeff Brower wrote: Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com wrote: Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards Again, iLBC is poor quality to begin with. You can't take a poor audio sample and make it better by converting it to a codec with better 'resolution'. An audio sample full of robot voice is going to sound like the same robot voice even if you transcode it to a better quality codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs. --Tim This just made me remember some comment on the iax.conf sample file... disallow=lpc10; Icky sound quality... Mr. Roboto. LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job with pitch detection so it tends to have a 'robotic' sound. With advent of MELPe, anyone needing bitrates 2400 or less should not be using LPC10. -Jeff MELPe is patent encumbered, so there is still a place for LPC10. LPC10 should sound a lot better than the one in Asterisk. The Asterisk codec is broken. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes
On 07/26/2010 11:57 AM, Alexander Aksarin wrote: On 20:59 Fri 23 Jul , Steve Underwood wrote: That's just how your images look for me, so I guess your problem is described here http://www.soft-switch.org/spandsp_faq/ar01s09.html Steve Big thanks for your help, Steve. I tried feh, gqview, gimp and pages look an odd shape. Can you say what image viewer you use for tiff? I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes
On 07/26/2010 10:55 PM, Tzafrir Cohen wrote: On Mon, Jul 26, 2010 at 09:54:24PM +0800, Steve Underwood wrote: On 07/26/2010 11:57 AM, Alexander Aksarin wrote: On 20:59 Fri 23 Jul , Steve Underwood wrote: That's just how your images look for me, so I guess your problem is described here http://www.soft-switch.org/spandsp_faq/ar01s09.html Steve Big thanks for your help, Steve. I tried feh, gqview, gimp and pages look an odd shape. Can you say what image viewer you use for tiff? I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. convert and the rest of imagemagick should handle multi-page tiff (e.g. convert it to PDF). The main value in converting FAX TIFFs to PDFs (which basically just encapsulates the TIFF file in a PDF wrapper) is that PDF readers generally get the images right. If the average image viewer was not so broken, converting FAXes to PDFs would be less popular. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes
On 07/23/2010 11:17 AM, Alexander Aksarin wrote: On 21:46 Thu 22 Jul , Steve Underwood wrote: It might help if you explained what you expect those pages should look like. I see three quite plausible pages. I expect to see this http://imagebin.ca/img/Eihpy0.jpg That's just how your images look for me, so I guess your problem is described here http://www.soft-switch.org/spandsp_faq/ar01s09.html Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk app_fax, T.30, weird received faxes
On 07/22/2010 12:15 PM, Alexander Aksarin wrote:
On 09:06 Thu 22 Jul , Alexander Aksarin wrote:
Hello to all. I have succesfully received fax by app_fax, but tif files are
weird.
There a faxes sended by several fax machines to asterisk.
http://filebin.ca/hnnumf/122.tif
http://filebin.ca/ospmn/151.tif
http://filebin.ca/fzuknc/151_.tif
Any ideas how to fix this?
debug log: http://filebin.ca/cashhg/full.today
part with fax from extensions.conf:
exten = fax,1,Goto(543,1)
exten = 543,1,Answer()
exten = 543,n,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}.tif)
exten =
543,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 543,n,Wait(3)
exten = 543,n,ReceiveFAX(${FAXFILE})
Some information about hardware:
Digium Wildcard TE110P T1/E1
fax machines:
Panasonic KX-FP153 // 151.tif
Panasonic KX-FT73 // 122.tif
scheme: fax- avayaT1 asterisk
Software:
OS: ALT Linux 5.0.1 Ark Server
asterisk 1.6.2.9
libspandsp6 0.0.6
It might help if you explained what you expect those pages should look
like. I see three quite plausible pages.
Steve
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Re: [asterisk-users] Is Centos 64 bit or 32 bit better?
On 06/29/2010 05:35 PM, Gareth Blades wrote: Zhang Shukun wrote: hi, all after a long time development, i need to deploy a production system. i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused me. my computer hardware support 64 bit OS. my question is : should i use Centos 5.4 64bit or Centos 5.4 32bit? which is better for my asterisk ? consider compatibilityand stability. this is a new machine , only used for asterisk, no other apps. Thank you in advance! 64bit will give you more adressible memory and faster performance when dealing with 64bit numbers. Neither of these will really give you any benefit but Asterisk and all the extras I have seen all work fine on 64bit so there is no real reason not to go for it. Actually most DSP code runs *much* faster on a 64 bit machine. I think it mostly the better register set which results in that. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDD/TTY Support
On 06/16/2010 11:31 PM, Karl Harris wrote: On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? There are installations where the TDD support in spandsp has been integrated with Asterisk, but I don't know if anyone has publicly released the code they use to integrate them. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDD/TTY Support
On 06/16/2010 11:44 PM, Danny Nicholas wrote: I’m supposing that it is 1. no better or worse than SMS support What relevance does SMS support have to TDD/TTY support? 1. dependent on the version you are on I don't think the TDD support has been touched for years, so I doubt the version makes much difference. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Karl Harris *Sent:* Wednesday, June 16, 2010 10:31 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] TDD/TTY Support On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? Thanks -- Karl Harris Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 aco1...@gmail.com wrote: Dear all, I've read that Asterisk supports only the G.729 A audio codec. I have several Grandstream IP phones with G.729 A/B codec implementation. Does G.729 A/B mean both version A and version B, or A/B is a new version different from A and B and it's not supported by Asterisk ??? G.729 is the base codec, which hardly anyone uses G.729 Annex A is a stripped down version which doesn't sound as good, but takes only half the compute power. This is the one almost everyone uses - who cares about voice quality, anyway? The bit stream is identical to G.729, so they are fully interworkable. For thos reason SDP does not distinguish between G.729 and G.729A. G.729 Annex B is a CNG/VAD add on for either of the above codecs. This feature may be turned on and off in the SDP, using the annexb parameter. A codec which cannot support Annex B is, therefore, always able to interwork with a codec that does support it. G.729AB or G.729A/B are the usual ways people described a codec which uses the Annex A version of the encoding and decoding, and which supports CNG/VAD. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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.
Asterisk always responds with a=T38MaxBitRate:2400.
I’ve tried with Patton and Grandstream devices and the result is always
the same.
Patton ignores the parameter and sends the fax at 9600.
Grandstream doesn’t, and all the faxes are going in and
out at 2400.
Looking at the code I found this in chan_sip.c (line 7736):
if ((sscanf(a, T38FaxMaxBuffer:%30u,x) == 1)) {
ast_debug(3, MaxBufferSize:%d\n, x);
found = TRUE;
} else if ((sscanf(a, T38MaxBitRate:%30u,x) == 1) ||
(sscanf(a, T38FaxMaxRate:%30u,x) == 1)) {
ast_debug(3, T38MaxBitRate: %d\n, x);
switch (x) {
case 14400:
p-t38.their_parms.rate = AST_T38_RATE_14400;
break;
case 12000:
p-t38.their_parms.rate = AST_T38_RATE_12000;
break;
case 9600:
p-t38.their_parms.rate = AST_T38_RATE_9600;
break;
case 7200:
p-t38.their_parms.rate = AST_T38_RATE_7200;
break;
case 4800:
p-t38.their_parms.rate = AST_T38_RATE_4800;
break;
case 2400:
p-t38.their_parms.rate = AST_T38_RATE_2400;
break;
}
found = TRUE;
else if {…
If I’m not misteaking the second “if else” condition will never be true
if the other device sends “T38FaxMaxBuffer” (wich they all usually do).
Shouldn’t it be
if((sscanf(a, T38FaxMaxBuffer:%30u,x) == 1) ((sscanf(a,
T38MaxBitRate:%30u,x) == 0) || (sscanf(a, T38FaxMaxRate:%30u,x)
== 0))) ??
No. You aren't understanding the code :-) It's comparing a string buffer
against various patterns, and the string can't match all the patterns at
the same time.
This code is executed as each line of the SDP is processed, and each one
will match one of the branches of this tree, and it's values will be
extracted and stored for later use.
In other words... this is not the cause of your problem.
Quite true, but the space in T38MaxBitRate: %d\n might be a problem,
as the number doesn't necessarily have a space in front of it.
Steve
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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 able to send over a 50 page fax from coast to coast with 0 issues However, did get this message in CLI: [May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:27:31] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:14] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:27] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found [May 13 07:28:39] WARNING[14700] app_fax.c: WARNING T.30 ECM carrier not found However, there was no noticeable errors in the fax., googling, the error didn't seem to make much since. This was via copper pair, over traditional LD carrier, into PRI terminating into a Sangoma card. Intel Xeon x3460, 8 gb ram, 320gb raid 0 sata Thanks to all who offered suggestions, and such, I will try this out, and hopefully should work well, as Steve Hinted to a year ago. William Stillwell Those messages mean exactly what they say - a chunk of image data was not decoded by the modem. The FAX protocol will retry the missing chunk of image, and by the end of the FAX you probably see no problems at all. The FAX will, however, taking somewhat longer than it should. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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) william.stillwell-li...@ablebody.net 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 they no longer offer their product to open source asterisk, only for there “WARP” appliance. NOT really looking to migrate from 1.4.x to 1.6.x So buy an asterisk appliance that supports fax, and then you can pay somebody else to do the upgrade. http://www.digium.com/en/products/appliance/ Native 1.6 fax is really quite good. It's worth reading the release notes and doing the upgrade. Does that appliance actually support FAX? The web pages don't mention it. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 asteriskl...@callthem.info mailto:asteriskl...@callthem.info wrote: On Thu, May 6, 2010 at 3:11 PM, Steve Totaro stot...@totarotechnologies.com mailto:stot...@totarotechnologies.com wrote: Yes, I purchased licenses for Fax for Asterisk and yes I called tech support and had the WORST experience I have ever had with any technical support call. I am running Asterisk 1.6.2.6 and: FAX For Asterisk Components: Applications: 1.6.2.0_1.2.0 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32) The guy was arrogant and absolutely a jerk and I don't like to call people names, but call it as I see it. This has not been my experience the five or six times I have had to call Digium over the years, but it has been many years since my last call so I have no idea what the general support staff is like. I could not get any questions answered by the tech that took hours to call me back to tell me to read the readme. That would be all well and good if I didn't pay money. He could not explain Digium's math as far as faxing and failed to offer to get back to me with any kind of answer. Maybe someone on the list can make sense of this Enron style of accounting: voipgw01*CLI fax show stats voipgw01*CLI FAX Statistics: --- Current Sessions : 1 Transmit Attempts: 0 Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 Digium G.711 Licensed Channels: 4 Max Concurrent : 1 Success : 0 Switched to T.38 : 0 Canceled : 0 No FAX : 1 Partial : 0 Negotiation Failed : 0 Train Failure: 3 Protocol Error : 0 IO Partial : 0 IO Fail : 0 voipgw01*CLI Digium T.38 Licensed Channels: 4 Max Concurrent : 4 Success : 175 Canceled : 0 No FAX : 6 Partial : 19 Negotiation Failed : 0 Train Failure: 83 Protocol Error : 33 IO Partial : 0 IO Fail : 0 Thanks, Steve Totaro wow definitely the acccounting engine is broken ... I can only make sense of this Receive Attempts : 336 Completed FAXes : 320 Failed FAXes : 57 1) your receive app was called 336 times but the fax hanged up before negotiating 2) you had 320 of this completed (partially or fully) 3) but 57 out of 320 failed to transmit entirely 57/320=17.8% which is too high for a commercial product IHMO Martin Considering that this is a direct cross connect from Leve3's cage to my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT VoIP service, I would expect nearly 100% success. Considering the circuit was just turned up and there is no data except Level3's phone traffic. They are our carrier, RespOrg, origination and termination, no 3rd parties, all on net. I could understand if it was a peaked out DIA circuit to some cut rate VoIP provider, but not under perfect circumstances. Thanks, Steve Totaro Were these all test calls made from a well defined source? It takes *two* correctly working FAX terminals to make a successful call. Its easy to get a high failure rate for silly reasons. In volume testing of spandsp and iaxmodem we had times where a high percentage of calls failed, which turned out to be just one rouge machine calling over and over again trying to achieve success. On the other hand, failures between known good FAX terminals should be far below 1%. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
