[Asterisk-Users] Re: Help with GPL license of Asterisk
>>>>> "Gerald" == Gerald Henriksen <[EMAIL PROTECTED]> writes: Gerald> On Thu, 02 Oct 2003 11:26:56 -0700, Jan Rychter Gerald> <[EMAIL PROTECTED]> wrote: >> Having worked with GPL software quite a bit, also in the commercial >> world, and having gotten some legal advice, I believe that the >> "anti-patent" clauses in the GPL and LGPL are quite possibly the >> biggest problem preventing the use of GPL'd software by commercial >> entities, much bigger than the "pass on the source and the rights" >> requirement. Gerald> Not really. Certainly it hasn't stopped lots of companies big Gerald> and small from releasing GPL software. >> As I understand it (and as my legal counsel advises me) this >> effectively means that if I distribute GPL/LGPL code, I have to make >> sure that its distribution and re-distribution is not restricted by >> patents (or other restrictions). Gerald> No, simply because that would be impossible (both because you Gerald> would never be able to program given the number of patents you Gerald> would have to search, and because it is entirely probable that Gerald> no software is entirely patent free). Gerald> What you can't do is knowingly license some source Gerald> code/software under the GPL/LGPL if you are already aware of Gerald> any patent or other issues that would in any way conflict with Gerald> the redistribution of that code. Where does it say "knowingly" in the GPL/LGPL text? I understand what you mean, I just think your interpretation of the GPL is different, and I do not agree with it. --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] Re: WCFXO echo rexolved for me
> "Brian" == Brian Schrock <[EMAIL PROTECTED]> writes: Brian> Hello, I resolved my echo issue using grandstream/estara etc etc Brian> sip phones and wcfxo interfaces from digium. I swapped out my Brian> via kt400 based msi kt4vl motherboard for an asus p4pe? i845? Brian> based motherboard and the echo has completly gone away along Brian> with aggressive suppressor option in the makefile. I hope this Brian> helps others. I'd guess this qualifies as a bug report -- is there a reason for X100P being so dependent on the motherboard used? Some of us don't have the option of swapping out the hardware. --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] Re: X100P too quiet
> "Ed" == Ed Dack <[EMAIL PROTECTED]>: Ed> I've got * up and running everything seems to work ok except for Ed> when you dial out using the X100P card. Ed> Everything sounds great this end but the person you call complains Ed> that they can't hear you very well (Very Whispered). Ed> Is their any way to turn up the volume. I've fiddled with the gain Ed> settings in the zapata.conf file but to no avail. Ed> Any help would be much appreciated! I have the same problem, except it seems to occur from time to time only (or only when calling specific numbers). I am also looking for a solution. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with GPL license of Asterisk
Mark Spencer: > The anti-patent clause was dropped ages ago. What do you mean? I can still see it in the LICENSE file in Asterisk. --J. > On Fri, 3 Oct 2003, Uriel Carrasquilla wrote: > > > So, is Astrisk being changed to an OSI-compliant license without the > > "anti-patent" clause? > > Uriel > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter > > Sent: Thursday, October 02, 2003 2:27 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Help with GPL license of Asterisk > > > > > > >>>>> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes: > > [...] > > Mark> No problem, it's easy to get confused :) I would, however, take > > Mark> issue with the GPL being "evil". It's not my *ideal* license, > > Mark> but it certainly is good enough. > > > > Just for the reference, while we're at it. GPL does have an issue, which > > can cause problems to some people or companies. It is often overlooked, > > because the "open source" issues seem much more controversial. > > > > Having worked with GPL software quite a bit, also in the commercial > > world, and having gotten some legal advice, I believe that the > > "anti-patent" clauses in the GPL and LGPL are quite possibly the biggest > > problem preventing the use of GPL'd software by commercial entities, > > much bigger than the "pass on the source and the rights" requirement. > > > > An excerpt from the GPL: > > > > 7. If, as a consequence of a court judgment or allegation of patent > >infringement or for any other reason (not limited to patent issues), > >conditions are imposed on you (whether by court order, agreement or > >otherwise) that contradict the conditions of this License, they do not > >excuse you from the conditions of this License. If you cannot > >distribute so as to satisfy simultaneously your obligations under this > >License and any other pertinent obligations, then as a consequence you > >may not distribute the Program at all. For example, if a patent > >license would not permit royalty-free redistribution of the Program by > >all those who receive copies directly or indirectly through you, then > >the only way you could satisfy both it and this License would be to > >refrain entirely from distribution of the Program. > > [...] > > 8. If the distribution and/or use of the Program is restricted in > >certain countries either by patents or by copyrighted interfaces, the > >original copyright holder who places the Program under this License > >may add an explicit geographical distribution limitation excluding > >those countries, so that distribution is permitted only in or among > >countries not thus excluded. In such case, this License incorporates > >the limitation as if written in the body of this License. > > > > As I understand it (and as my legal counsel advises me) this effectively > > means that if I distribute GPL/LGPL code, I have to make sure that its > > distribution and re-distribution is not restricted by patents (or other > > restrictions). > > > > If the code in question contains parts which some patents lay claim to, > > restricting distribution, then I must not distribute the code at > > all. Furthermore, by distributing the code I breach the GPL and expose > > myself to legal threat of a lawsuit from the FSF. > > > > It is needless to mention that it is impossible to me to verify that no > > patents (worldwide!) lay claim to the code I'm distributing and impose > > restrictions upon its distribution. Sooner or later I'm going to find > > out that I do not comply with the GPL, because I distribute GPLd code > > even though there are patent restrictions that apply to it. > > > > An example of a particularly clear case of this problem is the XviD code > > (http://www.xvid.org/), which is GPL-licensed. It seems to me that the > > authors (copyright holders, to be precise) may distribute the software > > under any license they choose, but nobody else is allowed to > > re-distribute it, because they would be violating section 7 of the GPL, > > as the MPEG-4 compression is (in some countries) covered by patents > > requiring royalties to be paid. > > > > This is an issue which is very often overlooked in the hot GPL > > debates. However, in the commercial world, it is possibly the most > > important one. > > > > Conclusion (IMHO of course): if you have the choice, use a license that > > is OSI-compliant but does not have the "anti-patent" clause. Or has it > > phrased differently. > > > > --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] echo for 15 seconds <002401c38308$2e05e0a0$0102010a@JUPITER>
>>>>> "Steve" == Steve Meyers <[EMAIL PROTECTED]> writes: Steve> On Thu, 2003-10-02 at 12:04, Jan Rychter wrote: >> I'm also hearing this, with an analog phone (connected to an >> S100U). Rather annoying. >> >> Incoming calls have an entirely different problem for me, a >> disastrous 5-8 second crackling/clicking sound, which seems to go >> quiet a while after you start speaking. The other side doesn't hear >> it, but it makes you miss the beginning of a call, e.g. you usually >> don't know who's calling :-/ This happens in a phone -> S100U -> * >> -> * -> X100P -> PSTN setup, when somebody is calling from the PSTN. Steve> The first server that I set up asterisk on had the same problem. Steve> I was using BudgeTones and a couple X100P's. Internal calls had Steve> no echo, etc, but calls over the X100P's had tons of echo for Steve> 10-15 sec. We also got a beeping sound. Steve> However, since the problem didn't seem widespread among X100P Steve> users, we decided it might be our server hardware, which while Steve> decent spec wise, was on the cheap end quality wise. We got Steve> some nicer hardware, and the problem went away. Any chance you could describe the hardware? Was it a Via-based board? I have a setup where I use two *'s, both on Via boards. One is a Mini-ITX and the other is a full-form motherboard. Would interrupt-sharing between the X100P and another card cause this problem? (there is simply no way to avoid it on some hardware!) --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Dutch PSTN-line on X100P
> "Eric" == Eric Wieling <[EMAIL PROTECTED]> writes: Eric> Check /proc/interrupts to make sure the cards are not shareing Eric> IRQs with anything. Is there anything that can be done so that this is not a requirement? There are (many) setups where this is simply not possible. Other cards can share interrupts -- why not those? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with GPL license of Asterisk
> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes: [...] Mark> No problem, it's easy to get confused :) I would, however, take Mark> issue with the GPL being "evil". It's not my *ideal* license, Mark> but it certainly is good enough. Just for the reference, while we're at it. GPL does have an issue, which can cause problems to some people or companies. It is often overlooked, because the "open source" issues seem much more controversial. Having worked with GPL software quite a bit, also in the commercial world, and having gotten some legal advice, I believe that the "anti-patent" clauses in the GPL and LGPL are quite possibly the biggest problem preventing the use of GPL'd software by commercial entities, much bigger than the "pass on the source and the rights" requirement. An excerpt from the GPL: 7. If, as a consequence of a court judgment or allegation of patent infringement or for any other reason (not limited to patent issues), conditions are imposed on you (whether by court order, agreement or otherwise) that contradict the conditions of this License, they do not excuse you from the conditions of this License. If you cannot distribute so as to satisfy simultaneously your obligations under this License and any other pertinent obligations, then as a consequence you may not distribute the Program at all. For example, if a patent license would not permit royalty-free redistribution of the Program by all those who receive copies directly or indirectly through you, then the only way you could satisfy both it and this License would be to refrain entirely from distribution of the Program. [...] 8. If the distribution and/or use of the Program is restricted in certain countries either by patents or by copyrighted interfaces, the original copyright holder who places the Program under this License may add an explicit geographical distribution limitation excluding those countries, so that distribution is permitted only in or among countries not thus excluded. In such case, this License incorporates the limitation as if written in the body of this License. As I understand it (and as my legal counsel advises me) this effectively means that if I distribute GPL/LGPL code, I have to make sure that its distribution and re-distribution is not restricted by patents (or other restrictions). If the code in question contains parts which some patents lay claim to, restricting distribution, then I must not distribute the code at all. Furthermore, by distributing the code I breach the GPL and expose myself to legal threat of a lawsuit from the FSF. It is needless to mention that it is impossible to me to verify that no patents (worldwide!) lay claim to the code I'm distributing and impose restrictions upon its distribution. Sooner or later I'm going to find out that I do not comply with the GPL, because I distribute GPLd code even though there are patent restrictions that apply to it. An example of a particularly clear case of this problem is the XviD code (http://www.xvid.org/), which is GPL-licensed. It seems to me that the authors (copyright holders, to be precise) may distribute the software under any license they choose, but nobody else is allowed to re-distribute it, because they would be violating section 7 of the GPL, as the MPEG-4 compression is (in some countries) covered by patents requiring royalties to be paid. This is an issue which is very often overlooked in the hot GPL debates. However, in the commercial world, it is possibly the most important one. Conclusion (IMHO of course): if you have the choice, use a license that is OSI-compliant but does not have the "anti-patent" clause. Or has it phrased differently. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configs for IAX <> IAX trunk
> "Brian" == Brian West <[EMAIL PROTECTED]> writes: Brian> Just a heads up.. you can't loop switch statements ie Brian> BOX A switch => BOX B BOX B switch => BOX A [...] I was actually wondering -- why? This is something I very naturally wanted to do the first time I configured two *'s. I wanted them to "exchange" dialplans, so that I don't have to replicate this information. I have some extensions on one of them, and others on the other, they are all unique and I want them all to be "globally" callable. So, why can't one do something like this? Is this a valid feature request? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] the g729 situation
> "LDM" == Louis-David Mitterrand <[EMAIL PROTECTED]> writes: LDM> Having purchased a license for 5 g729 channels on Digium's web LDM> shop I thought registration and installation would be a snap. NOT. LDM> I followed registration instructions to the letter but it failed LDM> with that message: LDM> ERROR! Your Internet connection is probably behind a proxy and the LDM> Registration program can't communicate with our server LDM> Which is stupid as my * box is a firewall and sits directly on the LDM> Internet whith no restrictions from in->out. I must say I'm impressed that people are brave enough to (1) accept the long, restrictive and sometimes outright scary (did you read the parts about credit card charges, or the definition of "G.729 software" in connection with "Improvement by Licensee"?) licensing agreement and (2) run a binary module that touches strange parts of the machine and communicates that information over the network to a third party. I also feel sorry for Digium, because they have to take the heat from unhappy users. IMHO this codec should be avoided at all cost. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds
> "Shaun" == Shaun Ewing <[EMAIL PROTECTED]> writes: Shaun> - Original Message - Shaun> From: Chad R. Graham >> For the first 15 seconds of a call I get echo on the ata 186 side >> only. I assume after that the echo canceller kicks in but is there >> any way to make it happen faster? Shaun> Same thing here - except we're using Cisco 7960 and 7940 IP Shaun> phones. Shaun> We're getting used to it, the main thing is that the remote Shaun> caller doesn't hear it (which they don't). A person visiting our Shaun> office and using the phone may get a bit of a surprise though. [...] I'm also hearing this, with an analog phone (connected to an S100U). Rather annoying. Incoming calls have an entirely different problem for me, a disastrous 5-8 second crackling/clicking sound, which seems to go quiet a while after you start speaking. The other side doesn't hear it, but it makes you miss the beginning of a call, e.g. you usually don't know who's calling :-/ This happens in a phone -> S100U -> * -> * -> X100P -> PSTN setup, when somebody is calling from the PSTN. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
> "Sean" == Sean P Robertson <[EMAIL PROTECTED]> writes: Sean> I have. Heads up on the built-in sound. Like everything else on Sean> the motherboard, it uses a VIA chipset and chan_oss will not work Sean> with it. Sean> Several posts have been made to the list in the past about the Sean> VIA chipset sound cards. Take a look at the Google archives for Sean> more info. Sean> Does anyone have any updated information on this or is the VIA Sean> chipset sound card a lost cause? Alsa works just fine for me on Via (8235 southbridge), although I didn't use it for Asterisk calls, only for recording audio. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] VIA vs Intel
> "Steven" == Steven Critchfield <[EMAIL PROTECTED]> writes: Steven> On Wed, 2003-09-24 at 13:13, Jon Pounder wrote: >> speaking of VIA - has anyone on the list looked at or used these ? >> http://www.mini-itx.com/store/default.asp?c=2¤cy=2 >> >> various collection of via based boards and cases and other goodies >> that go along with them. >> >> They are cheap enough they could work as either an asterisk server >> (diskless or with disk), or as phone platforms themselves. Steven> I was just looking at them since someone has built a mini Steven> distro to make one of these devices into a MythTV front end. I Steven> could see spending $200 per TV in my house to front them with Steven> these little boxes and then fill a couple machines up in a rack Steven> in the basement taping shows for the family. But, this is the Steven> wrong list to finish talking about this subject. I'm using the EPIA-M6000 with Asterisk. The only (serious) problem I have with it is that I'm unable to make the cards use the IRQs I want. I always get USB using the same interrupt as the X100P adapter, and the general mantra is that one should avoid that. If you know of a way to reassign interrupts in a saner manner, I'd appreciate any advice. Right now I have: 0:8540638 XT-PIC timer 1: 1201 XT-PIC keyboard 2: 0 XT-PIC cascade 10: 85365179 XT-PIC wcfxo, usb-uhci 11: 0 XT-PIC usb-uhci 14: 50857 XT-PIC ide0 15: 85732168 XT-PIC eth0, usb-uhci Otherwise, it's a fairly nice platform. You have to be careful to compile with proper flags, but otherwise it works very well. One thing: if you want to build a *quiet* PC, be careful. Many manufacturers' definitions of quiet will differ from yours. I've purchased an (overpriced) case from idot.com with a loud whining fan (louder than my PC!), which I've later exchanged for an (even more overpriced) case which still had a loud power supply fan. I ended up buying an $25 small power supply at Fry's and forcing that into the case. Obviously idot.com tests the "quietness" in a factory setting. Steven> As a phone platform, it may be overkill, but I bet it could Steven> drive a TDM400P card and be able to handle GSM compression. The Steven> question then again is if it is worth the cost for basically a Steven> 4 port asterisk based device like the ATA186? I use it with an X100P, and an S100U, with Speex and ILBC sometimes. I do have a number of problems, but they seem to be unrelated to the platform itself. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Very bad echo (appears that...)
> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes: Mark> I wouldn't mess with the gains if I were you. Mark What do you mean? Are the gains an unsupported feature? Aren't we supposed to adjust them? I have some people who complain that they can't hear me when I dial out using the X100P adapter. I thought that the gains were precisely for that purpose? --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Calls being interrupted, analog signalling problems
>>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes: Jan> I'm having trouble with a WX100USB adapter and a Siemens Gigaset Jan> cordless phone. Jan> If I select fxols as a signalling method, calls are being Jan> disconnected. Usually after about 4 minutes, and asterisk just Jan> says that the phone has hung up. Jan> If I choose fxogs, I immediately get a LINE IN USE message on my Jan> phone and I can't even get a dialtone. Jan> If I choose fxoks, it mostly works, but sometimes after making a Jan> call the adapter will get stuck in a LINE IN USE state, too. I Jan> don't know of a proper way to correct it, sometimes disconnecting Jan> the USB adapter and reloading the drivers and asterisk fixes that, Jan> sometimes not. Jan> What is the proper signalling method? What do you people use? I'd Jan> appreciate any advice. Hmm. Does the number of responses (zero) indicate that I'm the only one having such problems? I've just had to stop asterisk, unload the wcusb module, reload it, ztcfg it and start asterisk again, because my line was stuck in the LINE IN USE state (with no dialtone on the phone of course). I guess I'll report it as a bug, then. --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] Calls being interrupted, analog signalling problems
I'm having trouble with a WX100USB adapter and a Siemens Gigaset cordless phone. If I select fxols as a signalling method, calls are being disconnected. Usually after about 4 minutes, and asterisk just says that the phone has hung up. If I choose fxogs, I immediately get a LINE IN USE message on my phone and I can't even get a dialtone. If I choose fxoks, it mostly works, but sometimes after making a call the adapter will get stuck in a LINE IN USE state, too. I don't know of a proper way to correct it, sometimes disconnecting the USB adapter and reloading the drivers and asterisk fixes that, sometimes not. What is the proper signalling method? What do you people use? I'd appreciate any advice. --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 channel problems
> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]>: Jeremy> What part of "IN OTHER WORDS: Run Open H.323 v1.11.7, nothing Jeremy> newer, nothing older if u want this to work." don't you Jeremy> understand? Well, I was trying to find out (politely) about some things. Please allow me to paste from my previous E-mail: 1. >> Perhaps it's worth trying to report the bugs to distribution >> maintainers if indeed the distribution-specific installs of openh323 >> are this buggy? 2. >> Briefly, do I have a chance of reporting this bug with my versions >> of libraries, or is chan_h323 completely unsupported if I use >> anything other than 1.11.7? There was also an implicit question 3. Perhaps the docs haven't been updated and openh323 isn't this problematic anymore? You couldn't have answered question #2 any clearer. Also thanks to Brian West for his informative followup. --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] H.323 channel problems
I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the "Up" state, with asterisk consuming 100% of CPU: *CLI> show channels Channel (ContextExtensionPri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None) (None) 1 active channel(s) *CLI> show ch channel channels *CLI> show channel H323/ip$127.0.0.1:30008/21552 -- General -- Name: H323/ip$127.0.0.1:30008/21552 Type: H323 UniqueID: 1061946140.22 Caller ID: Jan <> DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 8 WriteFormat: 1024 ReadFormat: 1024 1st File Descriptor: 26 Frames in: 47575 Frames out: 94850 Time to Hangup: 0 -- PBX -- Context: local Extension: 123 Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (None) Stack: -1 Blocking in: ast_waitfor_nandfds *CLI> That's after hanging up (in gnomemeeting) on a H.323 call that is then bridged to IAX2. Now, before I go running to the bugtracker, I'd like to ask some general questions. The H.323 channel readme says: NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. And: Some chan_h323 users have reported success and others have reported dramatic failures when using newer versions of Open H.323. We haven't personally tested this and will not be able to assist you if you have 'issues'. Sorry. IN OTHER WORDS: Run Open H.323 v1.11.7 nothing newer nothing older if u want this to work. How does this relate to my bug? I'm using openh323-1.12 and pwlib-1.5.0 that I compiled myself. Do they have problems? Does this mean I am on my own? Perhaps it's worth trying to report the bugs to distribution maintainers if indeed the distribution-specific installs of openh323 are this buggy? The requirement of using this particular version of openh323 is a problem for those of us who also use other H.323 software (such as gnomemeeting) which specifically requires newer libraries. Briefly, do I have a chance of reporting this bug with my versions of libraries, or is chan_h323 completely unsupported if I use anything other than 1.11.7? many thanks, --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One-way audio using console
I've tried making calls using the console (both ALSA and OSS). ALSA seems to work after applying the little fix posted on this list some time ago by someone (which I'll submit into the bug tracker), but all I get is one-way audio: I can hear the other end, but nothing gets transmitted. At first I thought this was an audio problem, but it doesn't seem to be. My machine isn't transmitting audio (judging from what tcpdump says). I get pretty much the same effect using ALSA and using OSS on the asterisk side (ALSA OSS emulation on the kernel side, because my notebook only has ALSA). Am I going about this correctly? If I enter "dial " on the console, should I get a proper, full-duplex audio connection, or is there something else to be done? And a related question: entering "hangup" doesn't seem to do the right thing. I have to "soft hangup IAX2/..." and then "hangup" to get the desired effect. Is this how it is supposed to be? --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Open G.729A codec
> "Steve" == Steve Underwood <[EMAIL PROTECTED]>: Steve> Kim C. Callis wrote: >> I was reading on www.vovida.org/applications/downloads/G729A/ (home >> of VOCAL) pages, and that there is a free license use for >> non-commercial for G.729A. Is that usable under Asterisk or strictly >> a Vovida offering? >> Steve> This was a publicity stunt by VoiceAge, which Cisco/Vovida Steve> seemed to get dragged into in their determination to see G.729 Steve> become more widely used. All that ever really happened was a Steve> Windows binary was made available for very restricted use. This Windows binary is probably fairly easy to convert for someone with sufficient skills. It's a simple library, COFF format. It's probably sufficient to split it into .o files (using ar), then convert the .o files (using objcopy --target=elf32-i386, objcopy from cygwin has both elf32 and coff formats, so it's useful for that), and assemble the resulting elf32 .a library (again, using ar). What remains to be taken care of are mostly underscores in function/variable names. Otherwise, this process should work and one should be able to create a working Linux library (along with an asterisk codec). Just remember that this is for non-commercial, personal usage only, as the license clearly states. Also, one must not reverse-engineer the code, which the license prohibits. I was actually thinking about both buying a license for it and doing the above, to avoid the licensing monstrosity present in the G.729A codec resold by Digium. Then I gave it some thought and couldn't really find a reason to do so much work on non-free code while there was speex almost ready to be used. I think it is rather sad (not to say ridiculous) for a company to guard a piece of code this small with such monstrous licensing schemes. Steve> The G.729 implementation Digium supplies for Linux in from the Steve> same source. The licencing is so clunky I bet Mark is wishing he Steve> had left it alone! Couldn't agree more. The G.729 codec is so unDigium-like... don't buy it is my recommendation. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Open G.729A codec
> "Mark" == Mark Spencer <[EMAIL PROTECTED]>: >> This Windows binary is probably fairly easy to convert for someone >> with sufficient skills. It's a simple library, COFF format. It's >> probably sufficient to split it into .o files (using ar), then >> convert the .o files (using objcopy --target=elf32-i386, objcopy >> from cygwin has both elf32 and coff formats, so it's useful for >> that), and assemble the resulting elf32 .a library (again, using >> ar). What remains to be taken care of are mostly underscores in >> function/variable names. Mark> It's a little more complex than that. Remember the Windows one Mark> is single-channel only. It's not reentrant and thus totally Mark> useless for Asterisk unless you only need one channel. >> Otherwise, this process should work and one should be able to create >> a working Linux library (along with an asterisk codec). Mark> Which could not be distributed without violating GPL, nevermind Mark> Voicages licenses. [...] >> Just remember that this is for non-commercial, personal usage only, >> as the license clearly states. Also, one must not reverse-engineer >> the code, which the license prohibits. Mark> A requirement which you cannot apply to GPL'd code (unless you Mark> were the copyright holder as Digium is and thus able to make such Mark> exceptions). You are of course correct. I wasn't encouraging anyone to break licenses: what I was talking about was exactly single-channel personal use, no redistribution. Which just happened to be what I needed a while ago :-) >> Then I gave it some thought and couldn't really find a reason to do >> so much work on non-free code while there was speex almost ready to >> be used. Mark> Speex is really a great thing, but G.729 is the unfortunate Mark> standard for communicating with most (proprietary) SIP/H323 Mark> devices. If ATA 186's could talk SpeeX this wouldn't be a Mark> problem. Trying to get the Windows G.729 code ported to run with Mark> Asterisk is definitely barking up the wrong tree though, for both Mark> technical and legal reasons. BTW, I hope Speex support in Asterisk will get better. I still have some problems using it (the first several seconds of a call sound particularly bad). I did file a bug report and I'm waiting patiently. [...] >> Couldn't agree more. The G.729 codec is so unDigium-like... don't >> buy it is my recommendation. Mark> I don't think anybody buys G.729 just to have it. They buy it Mark> because they *have* to have it. And we sell it because they Mark> *have* to have it. I think eventually we'll be able to come up Mark> with a better (but not, for the near future, open) G.729 solution Mark> from us. I made a mistake of buying it so that I can have a low-bandwidth well-tested codec for use on an IAX2 link. Then I've caused Digium lots of unwanted trouble, because hair stood on the back of my neck after reading the licensing agreement and seeing the .so library. Let's hope it gets better in the future! --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] "Out of area" displayed as caller-id
When connecting an analog phone (Siemens Gigaset) to * via a WX100USB, the phone displays "Out of area" first, and then the caller id. The two displays alternate, making the caller-id hard to see. Is there any way I can tell the phone to just display the caller id? Out of area is a flag that gets sent during caller-id transmission, right? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 licensing -- an opinion
>>>>> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]>: Jeremy> Jan Rychter wrote: >> Please try to find a better solution. >> Jeremy> The DSP Group owns G.729. There is nothing anyone can do, they Jeremy> have us by the family jewels. We use iLBC and found it to be Jeremy> very acceptable in quality and bandwidth usage and its free. For what it's worth, I've settled on speex. It provides *excellent* quality, and very good bandwidth usage. It does have a few rough edges and doesn't work perfectly in Asterisk, but for personal use I find it just fine. I guess it could be problematic for larger installations, as it requires quite a bit of CPU time. I find it funny I've chosen a free codec over a commercial one even though money was not really an issue. --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] G.729 licensing -- an opinion
Seeing that many people here hit problems with activating their G.729 licenses, I decided to post my opinion. I have purchased two G.729 licenses, for my private use. I did this even though VoiceAge makes G.729 free for private use, as Windows libraries. I guess a sufficiently motivated person could take the COFF libraries, run them through objcopy on cygwin (producing ELF .o files) and link them and use for free under Linux. For personal use, of course, as any commercial usage still requires a valid license. I wasn't sufficiently motivated, so I just went ahead and purchased the licenses. Much to my surprise, after reading the monstrous licensing agreement presented on screen, I have discovered that: -- the definition of "Software" is broad enough to cover ANY G.729 software that I might ever be accessing or even *writing* myself, -- the "Improvement by Licensee" section is, well, rather "strange", -- the whole section 4 about "Payments by Credit Card" is something I am absolutely unwilling to agree to, having already paid for the license. This section has probably been left in by accident, but it exposes me to serious financial risk. Overall, I do not understand how anybody can expect me to run completely unknown binary software that even Digium says they don't know what it does, and which at a first glance: a) accesses files on my hard drive, b) accesses my SCSI devices, c) accesses my IDE devices, d) possibly accesses other devices via ioctl() calls, e) contains encryption code, f) possibly transmits sensitive information outside of my network. I have purchased a _codec_, which means encoding and decoding software. I did not expect any other functionality. I do not have the habit of running this sort of unknown "black-box" code on my machines. I'm not even mentioning the fact that the whole licensing is rather limited -- you can't move the license to another machine, and if you modify your hardware, your license will probably break, and you'll have downtime. I've found the licensing completely unacceptable, didn't accept the license, and asked for a refund, which Digium promptly granted. Now, I must stress that Digium has always been extremely nice and understanding, responded promptly and acted very fair. All of the business I have ever done with Digium went extremely well. Also, as the README file in the G.729 codec says -- the licensing is not really Digium code. It's VoiceAge stuff. Dear Digium! This piece of software is a disgrace. It really doesn't go together with the rest of what you are doing, which is of excellent quality. Please try to find a better solution. One minor suggestion for the immediate future: placing the full licensing agreement on the website would allow many people to read it before deciding on the purchase. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Speex support
> "John" == John Todd <[EMAIL PROTECTED]>: > "John" == John Todd <[EMAIL PROTECTED]> writes: > What is the state of speex support in asterisk? I saw the codec seems > to be there. John> Install the Speex library support, and re-compile Asterisk. John> There's probably a pre-compiled version of Speex for your system; John> look around in whatever package manager you use for your Linux John> distro. >> >> I do have the libraries installed. >> > Can speex be used on IAX2 links? Is there much work still to be done? >> John> Yes, it can be used. No work required to get functionality. >> >> Really? Have you tried it? I have. It doesn't work -- and a quick >> look at chan_iax2.c shows that there is a good reason for this -- >> get_samples() doesn't know how to calculate the number of samples >> for an incoming speex format frame. This results in chopped sound >> and hundreds of warnings: [snip] >> >> --J. PS: bad advice is worse than no advice... John> I take it that comment was directed at me. John> Yes, really, Speex does work, and yes, I did try it without any John> of the modifications you describe above. Feel free to ask for John> help if it doesn't work, but don't assume that others haven't John> made it work or that I'm giving you intentionally bad advice - John> it's insulting. I apologize, then -- I must have missed something, because after looking into it it seemed that there is no way it can work. I thought you just wrote without reading my posting carefully. sorry, --J. PS: I still think the patch I attached should be applied (or one that is more correct in calculating the number of samples), it made it work for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: list format vs newsgroup format
> "Steven" == Steven Critchfield <[EMAIL PROTECTED]> writes: Steven> On Fri, 2003-07-18 at 09:02, Chris Earle (CBL) wrote: >> Agh >> >> I hate trying to sift through all these messages and keep track of >> the various threads going on . >> >> Who else on here prefers the newsgroup/threaded approach? If you >> haven't already, check out news.gmane.org for mailing lists turned >> into newsgroups readable by news readers... Steven> What you need is to get a decent mail reader. Those of us that Steven> complain regularly about people changing subjects in the middle Steven> of a thread already know the benefits of threaded email Steven> reading. Why bother with a newsgroup because you choose to stay Steven> on windows and use outbreak express >> only problem being that this list requires list membership before >> posting Steven> And this is a good thing. Otherwise spammers only need the list Steven> address to spam us all, and you get this also on Steven> newsgroups. Right now the only risk is the fact that the email Steven> addresses we use here are archived publicly in an easy to Steven> harvest method. I think that is the most risk I wish to Steven> undertake. It's a real pity that the news.gmane.org interface is one-way only. GMANE makes my life a lot easier, I would never be able to subscribe to all those lists. asterisk-users is one of the very few lists that I am unable to properly access through GMANE. newsgroups are realy better than mailing lists, people. It's not a question of your mail reader (MUA), but of instantly available archives (press a key and access the entire history of a thread, even if you have just subscribed), bandwidth (I can't always afford to download 60 messages per day from asterisk-users when I'm on a GPRS link), and access (you get *instant* access to hundreds of mailing lists). As to spam, GMANE has an extremely neat spam-reporting and filtering solution, so I actually see almost on spam there. Please notice that in order to post to GMANE you have to prove you are not a spambot, so spam can come in from the mailing lists only. When people report it (single keypress in Gnus), it gets crossposted to gmane.spam.detected, so you can easily kill it based on the Xref: header. So, please don't judge GMANE by USENET standards, it's an entirely different thing. And it's a good thing. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Speex support
> "John" == John Todd <[EMAIL PROTECTED]> writes: >> What is the state of speex support in asterisk? I saw the codec >> seems to be there. John> Install the Speex library support, and re-compile Asterisk. John> There's probably a pre-compiled version of Speex for your system; John> look around in whatever package manager you use for your Linux John> distro. I do have the libraries installed. >> Can speex be used on IAX2 links? Is there much work still to be >> done? John> Yes, it can be used. No work required to get functionality. Really? Have you tried it? I have. It doesn't work -- and a quick look at chan_iax2.c shows that there is a good reason for this -- get_samples() doesn't know how to calculate the number of samples for an incoming speex format frame. This results in chopped sound and hundreds of warnings: WARNING[163851]: File chan_iax2.c, Line 605 (get_samples): Don't know how to calculate samples on 512 packets WARNING[163851]: File chan_iax2.c, Line 605 (get_samples): Don't know how to calculate samples on 512 packets WARNING[163851]: File chan_iax2.c, Line 605 (get_samples): Don't know how to calculate samples on 512 packets [...] [time passes] Ok, adding the following tiny modification to chan_iax2.c solves the problem: Index: chan_iax2.c === RCS file: /usr/cvsroot/asterisk/channels/chan_iax2.c,v retrieving revision 1.33 diff -u -r1.33 chan_iax2.c --- chan_iax2.c 16 Jul 2003 18:45:12 - 1.33 +++ chan_iax2.c 18 Jul 2003 19:16:49 - @@ -601,6 +601,9 @@ case AST_FORMAT_ADPCM: samples = f->datalen *2; break; + case AST_FORMAT_SPEEX: + samples = 160 * f->datalen; + break; default: ast_log(LOG_WARNING, "Don't know how to calculate samples on %d packets\n", f->subclass); } I don't know if that's correct, but I can now use speex on IAX2 links. It sounds considerably better than GSM. There is still one remaining problem which I do *not* know how to fix: when * plays messages from the hard disk stored in GSM format, I get choppy sound. It seems * can't properly deal with conversion from GSM to Speex. --J. PS: bad advice is worse than no advice... pgp0.pgp Description: PGP signature
[Asterisk-Users] Speex support
What is the state of speex support in asterisk? I saw the codec seems to be there. Can speex be used on IAX2 links? Is there much work still to be done? many thanks, --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
>>>>> "Matteo" == Brancaleoni Matteo <[EMAIL PROTECTED]> writes: Matteo> turn off jitterbuffer in both servers. aka jitterbuffer=no in Matteo> iax.conf Matteo> jitterbuffer, unfortunately, is buggy and don't work as Matteo> expected. Interesting -- this has indeed helped and the quality is better, too! But doesn't this mean I'm in trouble whenever the network decides to order packets around? --J. Matteo> Il mer, 2003-07-16 alle 20:45, Jan Rychter ha scritto: >> Hi, >> >> I'm running asterisk in the following setup >> >> Phone -> WX100USB -> * -> Internet -> * -> WX100P -> PSTN >> >> The two Asterisks talk to each other via IAX2 and use GSM for voice. >> >> This seems to work quite well except for occasional pauses in voice >> transmission. These seem to occur in _one_ direction only (when I'm >> on the phone, I can't hear the person that I called via the PSTN), >> last several seconds (as in one to five seconds) and are unrelated >> to network connectivity (a ping in another window runs just fine all >> the time). >> >> What could be the cause? What else could I do to help hunt down that >> bug? >> >> --J. ___ Asterisk-Users >> mailing list [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX pauses
Hi, I'm running asterisk in the following setup Phone -> WX100USB -> * -> Internet -> * -> WX100P -> PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 quality
Does G.729 provide better voice quality than GSM? (a question for people who have tried both) --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
>>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes: >>>>> "John" == John Todd <[EMAIL PROTECTED]> writes: John> For what it's worth, I have noticed the same problem, but I think John> the problem is in IAX2, since my long-haul portions of the John> diagram were over IAX2, while my SIP clients are almost always John> sitting on the same LAN as the Asterisk server. Jan> I have noticed these problems both in this kind of setup and in a Jan> SIP call to a remote Asterisk server. John> What codec were you testing with over IAX2? Jan> GSM. Jan> Having investigated this a bit more, it turns out that using alaw Jan> instead of gsm on the IAX2 link makes the problem go away. It Jan> seems the jitter settings start working then. Jan> Any hints? I'd prefer not to be stuck with 80kbps per call... Having investigated this further, it seems that connecting a zaptel device (WC100USB in my case) to the local * fixes the problem. --J. >> [I have sent a message about SIP problems via gmane, but it seems >> the list is gatewayed one-way only...] >> >> The message was: >> >> I've been trying to use Asterisk as a SIP->PSTN gateway. It runs >> fine when the SIP client is on the local network and there is not >> packet loss. But now I've tried running a remote client (halfway >> around the globe) -- this works great until some packets get >> lost. After that it seems that either my client (linphone) or >> Asterisk doesn't want to resynchronize -- what gets played back is >> all voice packets as they have been received. This creates an >> increasing lag in the conversation and the only way I've found to >> fix it is to disconnect and reconnect again. >> >> Is anyone else seeing this? Is it linphone's fault, or is it >> expected behavior? >> >> Now, I have tried running another * on "my" side of the link. The >> setup then becomes: >> >> linphone -> * -> internet (IAX2) -> * -> PSTN (or echo). >> >> I'm testing with the echo application (GSM used everywhere) and I'm >> getting the same thing: everything seems to work, but sooner or >> later there is an audio pause and the delay grows. It never gets >> back to normal. I've had it grow to as much as 10s. >> >> What makes it even more surprising is the network performance. I've >> had ping running in the background, same TOS settings, 10 packets >> per second. It shows that my RTT is (min/avg/max/mdev) >> 220/229/287/8.85 with 0% loss! That's a pretty good network. So >> where do the pauses and delays come from? >> >> --J. ___ Asterisk-Users >> mailing list [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users John> ___ Asterisk-Users John> mailing list [EMAIL PROTECTED] John> http://lists.digium.com/mailman/listinfo/asterisk-users Jan> ___ Asterisk-Users Jan> mailing list [EMAIL PROTECTED] Jan> http://lists.digium.com/mailman/listinfo/asterisk-users Jan> ___ Asterisk-Users Jan> mailing list [EMAIL PROTECTED] Jan> http://lists.digium.com/mailman/listinfo/asterisk-users pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] G729 licensing
> "Matthew" == Matthew Hardeman <[EMAIL PROTECTED]> writes: Matthew> I'm not familiar with the codec support in Gnomeeting, but Matthew> have you tried a codec like iLBC? I had great success running Matthew> ilbc over IAX2 between my home and office. It doesn't really matter all that much what I use in Gnomemeeting (or other client applications), as I have set up two *'s specifically for the purpose of transporting calls across the Internet. So as long as * is able to recode, I'm happy with any low-bandwidth codec that * supports. Which reminds me, I haven't been able to get recoding to work with the included h323 module. It always said "unable to find a path form 8 to 1" or similar. Is that a known issue, or should I investigate and report bugs? I'm using oh323 now, which seems to work. As for ILBC, it seems the quality is a bit too low. I need something of toll-quality, so that I don't have to explain to everyone where I'm calling from and why it sounds so strange. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] Using 2 PhoneJacks with Asterisk for Datacalls.
> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]> writes: Jeremy> Not after you've wasted the kind of money I did on that junk. Jeremy> I was even stupid enough to pay the extra $30 per card for Jeremy> G.729 and when I couldn't make it work on Linux, they told me Jeremy> it would never work on Linux due to licensing problems. Then Jeremy> they had the balls to tell me they had no facilities to refund Jeremy> the G.729 licensing fee's. I'd suggest to Digium to get their drivers into the standard Linux kernel. That's a fairly good form of advertising. Actually, now that I think of it, I also wanted to buy Quicknet hardware (because I saw it was supported under Linux), and then stumbled upon Digium accidentally. I am now a very happy user of one X100P, and a less happy user of a somethingUSB, which I haven't gotten to work yet. But the X100P was definitely worth the money. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] G729 licensing
> "Matthew" == Matthew Hardeman <[EMAIL PROTECTED]> writes: Matthew> Missing something? No... Matthew> So far as I'm aware there is no freely available G729 codec Matthew> available that will run under Linux... Kind of funny that Matthew> there *is* one for Windows, isn't it? Yes, puzzling. I guess one might go the way the other projects have (like mplayer or xine video players) -- use the Windows DLLs under Linux. This can be done with a bit of glue code. Matthew> As an aside, though, what kind of equipment are you using, and Matthew> what circumstances are you communicating in? ALAW & ULAW make Matthew> great codecs for use on a LAN. :) I'm using gnomemeeting (sometimes also linphone, but gnomemeeting is much better), asterisk with oh323 on one end, asterisk with X100P on the other end, doing the bridging to PSTN there. alaw and ulaw are all good and great, but the distance between the two asterisks is 18 hops and 9 hours of time difference, so I'd really like to save on the bandwidth. GSM would actually be fine if it wasn't for the sync problems that I've reported. --J. > Hi, > I'm looking for a good codec to use on a personal VoIP > setup. It is strictly for my personal use, I'll never resell > it, make money or it, or whatever. > It seems a free personal-use G729 codec is available as a > WIN32 library. I find it puzzling that at the same time one > has to pay license fees to use it under Linux, even > non-commercially. > I was wondering -- am I missing something? > --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
>>>>> "John" == John Todd <[EMAIL PROTECTED]> writes: John> This happens to me as I mention below, but only rarely. What is John> your CVS version? The latest? E.g. I've tested 2 days ago. --J. >> I'm curious. Isn't anyone else noticing these problems? Or are >> people simply not using asterisk for VoIP connectivity over >> wide-area networks this way? >> >> Or does it go away with g729 or other proprietary codecs? >> >> --J. >> > "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes: "John" == John Todd > <[EMAIL PROTECTED]> writes: John> For what it's worth, I have noticed the same problem, but I think John> the problem is in IAX2, since my long-haul portions of the John> diagram were over IAX2, while my SIP clients are almost always John> sitting on the same LAN as the Asterisk server. > Jan> I have noticed these problems both in this kind of setup and in a Jan> SIP call to a remote Asterisk server. > John> What codec were you testing with over IAX2? > Jan> GSM. > > Having investigated this a bit more, it turns out that using alaw > instead of gsm on the IAX2 link makes the problem go away. It seems > the jitter settings start working then. > > Any hints? I'd prefer not to be stuck with 80kbps per call... > > --J. > > [I have sent a message about SIP problems via gmane, but it seems the > list is gatewayed one-way only...] > > The message was: > > I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine > when the SIP client is on the local network and there is not packet > loss. But now I've tried running a remote client (halfway around the > globe) -- this works great until some packets get lost. After that it > seems that either my client (linphone) or Asterisk doesn't want to > resynchronize -- what gets played back is all voice packets as they > have been received. This creates an increasing lag in the > conversation and the only way I've found to fix it is to disconnect > and reconnect again. > > Is anyone else seeing this? Is it linphone's fault, or is it expected > behavior? > > Now, I have tried running another * on "my" side of the link. The > setup then becomes: > > linphone -> * -> internet (IAX2) -> * -> PSTN (or echo). > > I'm testing with the echo application (GSM used everywhere) and I'm > getting the same thing: everything seems to work, but sooner or later > there is an audio pause and the delay grows. It never gets back to > normal. I've had it grow to as much as 10s. > > What makes it even more surprising is the network performance. I've > had ping running in the background, same TOS settings, 10 packets per > second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 > with 0% loss! That's a pretty good network. So where do the pauses > and delays come from? > > --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
I'm curious. Isn't anyone else noticing these problems? Or are people simply not using asterisk for VoIP connectivity over wide-area networks this way? Or does it go away with g729 or other proprietary codecs? --J. > >>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes: > >>>>> "John" == John Todd <[EMAIL PROTECTED]> writes: > John> For what it's worth, I have noticed the same problem, but I think > John> the problem is in IAX2, since my long-haul portions of the > John> diagram were over IAX2, while my SIP clients are almost always > John> sitting on the same LAN as the Asterisk server. > > Jan> I have noticed these problems both in this kind of setup and in a > Jan> SIP call to a remote Asterisk server. > > John> What codec were you testing with over IAX2? > > Jan> GSM. > > Having investigated this a bit more, it turns out that using alaw > instead of gsm on the IAX2 link makes the problem go away. It seems the > jitter settings start working then. > > Any hints? I'd prefer not to be stuck with 80kbps per call... > > --J. > > > [I have sent a message about SIP problems via gmane, but it seems the > > list is gatewayed one-way only...] > > > > The message was: > > > > I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine > > when the SIP client is on the local network and there is not packet > > loss. But now I've tried running a remote client (halfway around the > > globe) -- this works great until some packets get lost. After that it > > seems that either my client (linphone) or Asterisk doesn't want to > > resynchronize -- what gets played back is all voice packets as they > > have been received. This creates an increasing lag in the > > conversation and the only way I've found to fix it is to disconnect > > and reconnect again. > > > > Is anyone else seeing this? Is it linphone's fault, or is it expected > > behavior? > > > > Now, I have tried running another * on "my" side of the link. The > > setup then becomes: > > > > linphone -> * -> internet (IAX2) -> * -> PSTN (or echo). > > > > I'm testing with the echo application (GSM used everywhere) and I'm > > getting the same thing: everything seems to work, but sooner or later > > there is an audio pause and the delay grows. It never gets back to > > normal. I've had it grow to as much as 10s. > > > > What makes it even more surprising is the network performance. I've > > had ping running in the background, same TOS settings, 10 packets per > > second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 > > with 0% loss! That's a pretty good network. So where do the pauses > > and delays come from? > > > > --J. pgp0.pgp Description: PGP signature
[Asterisk-Users] G729 licensing
Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something? --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] audio pause/delay problems
>>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes: >>>>> "John" == John Todd <[EMAIL PROTECTED]> writes: John> For what it's worth, I have noticed the same problem, but I think John> the problem is in IAX2, since my long-haul portions of the John> diagram were over IAX2, while my SIP clients are almost always John> sitting on the same LAN as the Asterisk server. Jan> I have noticed these problems both in this kind of setup and in a Jan> SIP call to a remote Asterisk server. John> What codec were you testing with over IAX2? Jan> GSM. Having investigated this a bit more, it turns out that using alaw instead of gsm on the IAX2 link makes the problem go away. It seems the jitter settings start working then. Any hints? I'd prefer not to be stuck with 80kbps per call... --J. > [I have sent a message about SIP problems via gmane, but it seems the > list is gatewayed one-way only...] > > The message was: > > I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine > when the SIP client is on the local network and there is not packet > loss. But now I've tried running a remote client (halfway around the > globe) -- this works great until some packets get lost. After that it > seems that either my client (linphone) or Asterisk doesn't want to > resynchronize -- what gets played back is all voice packets as they > have been received. This creates an increasing lag in the > conversation and the only way I've found to fix it is to disconnect > and reconnect again. > > Is anyone else seeing this? Is it linphone's fault, or is it expected > behavior? > > Now, I have tried running another * on "my" side of the link. The > setup then becomes: > > linphone -> * -> internet (IAX2) -> * -> PSTN (or echo). > > I'm testing with the echo application (GSM used everywhere) and I'm > getting the same thing: everything seems to work, but sooner or later > there is an audio pause and the delay grows. It never gets back to > normal. I've had it grow to as much as 10s. > > What makes it even more surprising is the network performance. I've > had ping running in the background, same TOS settings, 10 packets per > second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 > with 0% loss! That's a pretty good network. So where do the pauses > and delays come from? > > --J. ___ Asterisk-Users > mailing list [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users John> ___ Asterisk-Users John> mailing list [EMAIL PROTECTED] John> http://lists.digium.com/mailman/listinfo/asterisk-users Jan> ___ Asterisk-Users Jan> mailing list [EMAIL PROTECTED] Jan> http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] audio pause/delay problems
> "John" == John Todd <[EMAIL PROTECTED]> writes: John> For what it's worth, I have noticed the same problem, but I think John> the problem is in IAX2, since my long-haul portions of the John> diagram were over IAX2, while my SIP clients are almost always John> sitting on the same LAN as the Asterisk server. I have noticed these problems both in this kind of setup and in a SIP call to a remote Asterisk server. John> What codec were you testing with over IAX2? GSM. --J. >> [I have sent a message about SIP problems via gmane, but it seems >> the list is gatewayed one-way only...] >> >> The message was: >> >> I've been trying to use Asterisk as a SIP->PSTN gateway. It runs >> fine when the SIP client is on the local network and there is not >> packet loss. But now I've tried running a remote client (halfway >> around the globe) -- this works great until some packets get >> lost. After that it seems that either my client (linphone) or >> Asterisk doesn't want to resynchronize -- what gets played back is >> all voice packets as they have been received. This creates an >> increasing lag in the conversation and the only way I've found to >> fix it is to disconnect and reconnect again. >> >> Is anyone else seeing this? Is it linphone's fault, or is it >> expected behavior? >> >> Now, I have tried running another * on "my" side of the link. The >> setup then becomes: >> >> linphone -> * -> internet (IAX2) -> * -> PSTN (or echo). >> >> I'm testing with the echo application (GSM used everywhere) and I'm >> getting the same thing: everything seems to work, but sooner or >> later there is an audio pause and the delay grows. It never gets >> back to normal. I've had it grow to as much as 10s. >> >> What makes it even more surprising is the network performance. I've >> had ping running in the background, same TOS settings, 10 packets >> per second. It shows that my RTT is (min/avg/max/mdev) >> 220/229/287/8.85 with 0% loss! That's a pretty good network. So >> where do the pauses and delays come from? >> >> --J. ___ Asterisk-Users >> mailing list [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users John> ___ Asterisk-Users John> mailing list [EMAIL PROTECTED] John> http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] audio pause/delay problems
[I have sent a message about SIP problems via gmane, but it seems the list is gatewayed one-way only...] The message was: I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine when the SIP client is on the local network and there is not packet loss. But now I've tried running a remote client (halfway around the globe) -- this works great until some packets get lost. After that it seems that either my client (linphone) or Asterisk doesn't want to resynchronize -- what gets played back is all voice packets as they have been received. This creates an increasing lag in the conversation and the only way I've found to fix it is to disconnect and reconnect again. Is anyone else seeing this? Is it linphone's fault, or is it expected behavior? Now, I have tried running another * on "my" side of the link. The setup then becomes: linphone -> * -> internet (IAX2) -> * -> PSTN (or echo). I'm testing with the echo application (GSM used everywhere) and I'm getting the same thing: everything seems to work, but sooner or later there is an audio pause and the delay grows. It never gets back to normal. I've had it grow to as much as 10s. What makes it even more surprising is the network performance. I've had ping running in the background, same TOS settings, 10 packets per second. It shows that my RTT is (min/avg/max/mdev) 220/229/287/8.85 with 0% loss! That's a pretty good network. So where do the pauses and delays come from? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
