RE: [Asterisk-Users] The Smallest Asterisk Server Ever?
It's noteworthy that while Linux is GPL'ed, that doesn't mean that the userspace applications that essentially make up the Snom phone and run on top of the GPL'ed Linux kernel are. Snom will gladly give you their customizations to the kernel, and a build environment that will produce a firmware image that can be loaded onto the Snom's... They will not, however, unless policy has changed, give you the source to the phone itself... I tried. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, February 03, 2004 12:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? On Tue, 2004-02-03 at 12:01, Chris Albertson wrote: I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. Similarly, I know there was a stink about Linksys using linux inside a router. I just picked up a USR 802.11g router that would be cool to get a small VoIP only asterisk install on. It would make setting up those 802.11b phones nice and easy. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] consultative transfer cisco
Yes. It works as it is supposed to. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Thursday, October 16, 2003 1:31 PM To: ASTERISK USERS Subject: [Asterisk-Users] consultative transfer cisco Hello, Is it possible to makeconsultative transfer on Cisco 7940 and 7960 phones? -- Bart
RE: [Asterisk-Users] Cisco CallManager Image for 7940/7960
Dustin, It's quite a pain to get those without a CallManager... However, there are some tools for extracting the compressed files from an InstallShield image and I have successfully done so with those files in particular and was able with some tweaking to get a phone back to Skinny without having a CallManager. Good luck. If you need a pointer or two, drop me a line at [EMAIL PROTECTED] Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of DUSTIN WILDES Sent: Friday, October 03, 2003 12:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco CallManager Image for 7940/7960 Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have). ___ 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
RE: [Asterisk-Users] SIP security (was: New ATA clone out)
I've never attempted this in CIPE, though I have tested a SIP session through IPSEC with no trouble. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Monday, September 29, 2003 2:28 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP security (was: New ATA clone out) Is it not possible to bundle all of the TCP/UDP traffic in a CIPE tunnel (or similar). has anyone tried it? what were the results? senad ___ 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
RE: [Asterisk-Users] Nortel M Series phones support
Nope... Nortel's M series is for the Meridian series PBXs/switches... It's a proprietary digital interface. I doubt it will ever happen. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason A. Pattie Sent: Monday, September 29, 2003 4:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Nortel M Series phones support -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've searched the mailing list quite extensively, but didn't come up with anything promising (some things wer helpful, though). Does anyone know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310) phones can be made to work with the TDM400P card or if they are ADSI compatible at all? I kind of doubt they will work if they are not compatible, but I don't know what it would take to plug them directly into a * box. Thanks. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/eKEcuYsUrHkpYtARAtP0AJ9LKw5RucJpsTgp7/nq9yVGJt24JACeLuQP OcXnVL7jnqRwXD2VNX/SiC0= =rAM3 -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Mailscanner thanks transtec Computers for their support. ___ 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
RE: [Asterisk-Users] CDR Web Search Frontend
I doubt if you'll be able to. They made a strategic decision to GPL rather than LGPL their client access libraries, as they wanted to up their proprietary license revenue. Essentially, they're trying to enhance the benefits of paying for a commercial license fee, by making it difficult to use MySQL in non-GPL products without said license. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Monday, September 29, 2003 4:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CDR Web Search Frontend Personally, I *love* MySQL, and I'm a bit surprised by their sudden change from public domain (and maybe LGPL) to GPL for their client libraries... Who can we bug at mysql to see if we can get that changed? bkw ___ 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
Re: [Asterisk-Users] G729 experiences..
The last time I checked, the Snom 200's were so out-tasked by the G.729 implementation on them that their U.I. became non-responsive during the call... Serious performance issues there. They may have come up with a clever fix, but it's been a while since I've looked into it... At the time, what I saw was scary. Matt Hardeman PaperSoft - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 26, 2003 4:43 AM Subject: Re: [Asterisk-Users] G729 experiences.. Matthew Hardeman wrote: It's ok... The voice sounds fine. It's superior to most cell phone calls, anyway. I've used it with the Cisco 7960's without any trouble. You can use asterisk in any way that uses it in console mode. Safe asterisk does so, so you can use it. This may be otherwise fixed, but I'm not sure. As safe asterisk works, I don't worry about it. Voicemail will use one license for each output stream it has to transcode. Therefore, it is preferable if you are using G729 to only write out one format of voicemail recording. I use WAV49, which is small like GSM, but easier to play on default windows installs with any kind of decent media player installed. It *does* properly release the license when done. (At least now, on my system, it does.) Matt Hardeman PaperSoft Has anyone used the Digim G.729 codec with SNOM 200 phones? I have heard people have had success with the Grandstream phones.. ___ 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
RE: [Asterisk-Users] G729 experiences..
It's ok... The voice sounds fine. It's superior to most cell phone calls, anyway. I've used it with the Cisco 7960's without any trouble. You can use asterisk in any way that uses it in console mode. Safe asterisk does so, so you can use it. This may be otherwise fixed, but I'm not sure. As safe asterisk works, I don't worry about it. Voicemail will use one license for each output stream it has to transcode. Therefore, it is preferable if you are using G729 to only write out one format of voicemail recording. I use WAV49, which is small like GSM, but easier to play on default windows installs with any kind of decent media player installed. It *does* properly release the license when done. (At least now, on my system, it does.) Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Thursday, September 25, 2003 7:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 experiences.. Hi, I am still toying with the idea of going ahead with using the G.729.. Can those using it tell me about some of your experiences using G.729.. Things like and problems you had running it, the voice quality and anything else you can think of... I have read in the archives that asterisk has to be run with -c.. Is this still the case? and if so does this mean that * can't be run using the safe_asterisk script? or started remotely via an SSH session?? I have also read that the voicemailmain app uses up licences.. Does this still happen and how many does it use?? Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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
Re: [Asterisk-Users] VoIP Support for Symbian OS Devices
It can be done, with some limitations. I've tried a couple simple tests between a couple of Nokia 3650's... It's noteworthy that the fastchat program for symbian phones has some voip capabilities... See www.fastchat.com for more info. As far as trying to do a true, realtime bi-directional voip streaming from a symbian phone... Don't bother... GPRS has exceptionally high round trip latency, averaging over 700ms generally and sometimes even more than one second. SSH is painful over GPRS, two-way voip would be unbearable. Matt Hardeman PaperSoft - Original Message - From: Andrew Joakimsen To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 9:37 PM Subject: [Asterisk-Users] VoIP Support for Symbian OS Devices Does anyone have any insignt on this? Any client programs that could be used?
RE: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware
I've found there are some bugs they don't list (bugs of great severity) that are fixed in the latest release that can cause trouble in certain environments. The phone's handling of ICMP redirects in a multihomed Ethernet environment (two separate, exclusive subnets running on a single segment) is very flaky, and frequently will result in the phone crashing and rebooting. The latest release seems to mitigate this in most instances, though I actually found the best solution (sadly) was to prevent my gateway from sending ICMP redirects to the Cisco phones. The 5.x+ stuff is annoying, due to the whole code-signing issue. It's kind of anti-open-architecture... On the other hand, there aren't any non-Cisco firmware builds for these phones floating around out there... Would you even want one? Cisco implemented the code signing enforcement as a response to a security analysis of the phones that pointed to the ability to make the phone run arbitrary code via TFTP being a security risk. That risk is no longer. I have mixed feelings about it, but have no regrets in having deployed the 5.x solutions in my business. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, September 23, 2003 10:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Advantage of Cisco 7960 with 5.x firmware their really isn't much fixed between 4.4 and the 5.x stuff but at the time thats all I had. So I put that on the phone. So far everything works like a champ. Not one problem. 4.4 http://www.cisco.com/en/US/products/sw/voicesw/ps2156/prod_release_note0 9186a008016096f.html#63943 Resolved Caveats.Release 5.0 No resolved caveats specific to Cisco IP Phone 7940/7960 Release 5.0 require documentation in these release notes. Resolved Caveats.Release 5.1 All caveats listed in this section are resolved in Cisco IP Phone 7940/7960 Release 5.1. This section lists only severity 1 and 2 caveats and select severity 3 caveats CSCdz59328: SIPPhone: The UI responsiveness slow, fast fingers cause digit drop CSCdz77783: SIPPhone: Clipping of voice in 7960 SIP phone CSCea83100: SIP: Dialing # does not work correctly if dialplan is empty CSCea85697: Phone may fail to reset when an Exception occurs CSCea93250: SIPPhone: Dialing # does not always work if default rule missing CSCeb27906: SIPPhone: Null To-tag in REFER causes transfer fail (race condition) CSCeb29575: SIPPhone: NOTIFY Event header shortform is not supported (o:) Resolved Caveats.Release 5.2 All caveats listed in this section are resolved in Cisco IP Phone 7940/7960 Release 5.2. This section lists only severity 1 and 2 caveats and select severity 3 caveats CSCeb41335: DSP mismatch with upgrade failure CSCeb44769: Phone removes dots in the IP address when sending ACK CSCeb46028: 79x0 Memory leak issues related to DNS query failures CSCeb75975: Phone crashes upon any menu exit Resolved Caveats.Release 5.3 All caveats listed in this section are resolved in Cisco IP Phone 7940/7960 Release 5.3. This section lists only severity 1 and 2 caveats and select severity 3 caveats CSCeb85936: SIP phone doesnt use the medium level contact field Hope that helps. bkw On Tue, 23 Sep 2003, Peter Pauly wrote: I'm currently running firmware version 3.2 on my Cisco 7960. I've seen on the list that several people are running the 5.x latest versions. I've avoided going to higher firmware versions because I'm worried about potential problems or issues with the encryption mechanism used in the later firmware versions. (Once you go to an encrypted firmware version, you can't go back, right?) For those of you who have gone to the newer firmware, what features or benefits have you seen by going with the newer firmware? ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Source for 50-pin amphenol cables?
GrayBar or any electrical supply that handles telephony has them. Matt Hardeman PaperSoft - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 13, 2003 10:00 AM Subject: Re: [Asterisk-Users] Source for 50-pin amphenol cables? On Sat, 2003-09-13 at 09:18, Peter Pauly wrote: I'm looking for a source for 50-pin amphenol cables, the ones used to connect Adtran's to punch down blocks. Preferably, one that's mail order and takes orders over the internet. Thanks. You should be able to get it at any decent size contractor store. Also any company that does a lot of cable sales. BTW, this is a good time to point out that Google has launched some very good services that let you do better searches when you want to purchase something rather than read about it. You should check out http://www.froogle.com/ When I went to froogle, I search for 25 pair telephone. From there there was a advertiser that linked me to this URL. http://www.cablesamerica.com/product_list.asp?cat%5Fid=1305 While I am reminding everyone why google rocks and you should look there often, You should also check out the http://catalogs.google.com They have several really interesting catalogs that you can search through. It has helped me find information on parts where those types of parts are not yet available via web pages. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
RE: [Asterisk-Users] Cisco 7960 + SIP
To clarify, you have to upgrade to 3.1 or 3.2 first, before attempting a version four load. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Sent: Friday, September 12, 2003 1:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 + SIP [EMAIL PROTECTED] tftpboot]# cat OS79XX.TXT P0S30100 Get this image as well. Shaun Ewing wrote: Hello all, I know this isn't strictly Asterisk, but I'm sure that there are more people here using the Cisco 7960 w/ SIP, so I thought I'd post here. I've just bought a Cisco 7960 phone to use with Asterisk. It came with the CallManager image on it. I've got the 4.4 SIP images (P0S3-04-4-00). If I put P0S3-04-4-00 in the OS79XX.TXT file, the phone downloads this fine (watching TFTP server debug). It then proceeds to request P0S3-04-.bin. I don't know why. Naturally this file isn't found. I tried renaming the file to P0S3-04-.bin. The phone then downloads around 80% before aborting. I hope somebody might be able to shed some light on the situation. Any help would be greatly appreciated. Thanks, Shaun ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G729
Yes! The registration hardware locks to system parameters, and does some software locking to certain inodes in your linux fs. You must run registration on the production server. Their license scheme sucks! Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zac Sprackett Sent: Friday, September 12, 2003 3:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 Hi, I recently purchased some G729 licenses for asterisk. I'm concerned with the registration process. My build tools are not physically located on the same machine from which I build asterisk. I build RPMs on another machine and then install them on my production server. Am I going to cause myself trouble by runnning Registration on my non build host? Thanks -z ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Open G.729A codec
Another approach would be... Just modify the mod_g729b.so such that the licensing constraints aren't so problematic... A little bird said it shouldn't be hard to do so... Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Rychter Sent: Tuesday, August 12, 2003 12:08 PM To: [EMAIL PROTECTED] Subject: 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. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP phone recommendation
I've never actually placed a call with the budgetone, but it just looks and feels cheap. The Snom is much nicer, and I tend to find that the Cisco 7960s/7940s are quite nice as well. Matt Hardeman PaperSoft - Original Message - From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 12, 2003 10:12 PM Subject: RE: [Asterisk-Users] IP phone recommendation How about when you compare the SNOM to the Budgetone, which one would you recommend for basic telephony? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Tuesday, August 12, 2003 2:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IP phone recommendation I wasn't refering to the costs of things on ebay.. I was talking about new prices.. Hell you could get a Ferrari on ebay for 20 bucks if you are really lucky.. :) Later.. On Tue, 2003-08-12 at 11:45, WipeOut . wrote: The Cisco is from what I have heard a good phone but is VERY expenisve.. My suggestions would be to go with either a SNOM 200 or a Grandstream Bugetone.. Where can one get a SNOM 200 for less than a Cisco 7960? The Cisco's are about $300 on eBay (with power supply). I can't find a SNOM 200 on eBay, and retail seems to be $300. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Open G.729A codec
If I had to venture a guess, I would say that the protection scheme is in place in the hopes that everyone will use their implementation rather than reinvent the wheel. If this is indeed the case, their protection scheme is useful in helping to protect the patent license as well as their code. So far, it would seem, no one has bothered to reinvent the wheel, and as such we're stuck using their implementation. Matt Hardeman PaperSoft - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 12, 2003 8:30 PM Subject: Re: [Asterisk-Users] Open G.729A codec Eric Wieling wrote: On Tue, 2003-08-12 at 15:37, Mark Spencer wrote: Couldn't agree more. The G.729 codec is so unDigium-like... don't buy it is my recommendation. I don't think anybody buys G.729 just to have it. They buy it because they *have* to have it. And we sell it because they *have* to have it. I think eventually we'll be able to come up with a better (but not, for the near future, open) G.729 solution from us. What is the license for? The actual binary module or for the patented codec? If it's for the codec, then why can't you get a license from voiceage and then use your own code. As you said it's available from the ITU. I have no idea why VoiceAge want to protect the code as they do. The code isn't interesting to licence. Its the pool of patents you really need to licence, and that is bundled with the VoiceAge codec. I don't know if they indemnify their licencees with regard to other patent holders crawling out of the woodwork with fresh claims on G.729, but they do include a licence for the known patents. Believe VoiceAge have some kind of exclusive pool licencing rights. I'm not clear how this works, though. The ITU G.729 code is pretty much useless for real world use. It is very slow. It gets the right answers, but not by efficient means. All the voice codec reference code I have seen is like this. The people who develop these things *have* to write an efficient version, as standards bodies demand to know the approximate MIPS a good implementation will require. The implementors do not release this version as the reference model. I've been through this from the codec developer's side. The reference model may be 10 or more times slower than a commercial grade implementation. I've no idea what the ratio might be for G.729. If someone produced a good open implementation of G.729, then it might be interesting to see how much the patents could be licenced for. The usual problem with these pooling things is they offer you two deals: One is US$many per port for one port up. The other is US$little per port for larger volumes, but you need to pay a one off fee of US$100,000 (or something on that scale) up front. Regards, Steve ___ 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
Re: [Asterisk-Users] Open G.729A codec
I completely see your point, and I agree with you that sales of the item would be much higher if they didn't have their silly scheme. However, it seems to me that they intend to jealously defend and over-enforce... Having said that, one could surmise that they are simply control freaks hoping no one will start selling a better implementation. For that matter, is it possible that they've used their patent positions to discourage others from trying to build another implementation? One must wonder why someone hasn't written and started distributing one? Matt Hardeman PaperSoft - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 12, 2003 9:22 PM Subject: Re: [Asterisk-Users] Open G.729A codec Hi Matthew, That argument doesn't seem to work. I don't hear many complaints here about the cost of the VoiceAge codec. It's the clunkiness of the protection scheme people don't like. It's only the protection scheme that seems to be making people want to dump the VoiceAge code. Remember how Microsoft got to be so big? Most successful packages, like 123, had clunky copy protection that hurt the genuine customers far more than the pirates. Microsoft's applications business was getting nowhere at that time. Then Microsoft make a big announcement that they would not use such clunky protection schemes on Word or Excel, and their applications sales have never looked back. Inconveniencing the genuine customers is a proven loser. Perhaps the music industry will learn this soon. Regards, Steve Matthew Hardeman wrote: If I had to venture a guess, I would say that the protection scheme is in place in the hopes that everyone will use their implementation rather than reinvent the wheel. If this is indeed the case, their protection scheme is useful in helping to protect the patent license as well as their code. So far, it would seem, no one has bothered to reinvent the wheel, and as such we're stuck using their implementation. ___ 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
Re: [Asterisk-Users] Open G.729A codec
Are the VoiceAge people generally unpleasant to work with and geniunely uncaring, or do they just fail to respond? Matt Hardeman PaperSoft - Original Message - From: Mark Spencer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 12, 2003 10:16 PM Subject: Re: [Asterisk-Users] Open G.729A codec 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! Believe it or not, we worked hard to get that license agreement *improved*. I wish they took our concerns (and those of our customers) more seriously. Mark ___ 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
RE: [Asterisk-Users] (no subject)
I've seen a number of anti-spam services that functioned on a community basis (automatically) by accepting submissions from verifiably distinct users and only blocking temporarily (for exponentially increasing time intervals) based on an algorithm that factored in unique reports from distinct users and the trust level of said user... A concept like that would be fun to try, but I'm not sure anyone would be willing to do it to their production systems. :) Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Tuesday, August 05, 2003 4:13 PM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] (no subject) What if someone adds your number to that list ? Someone would have to moderate it. regards Martin On Tue, 5 Aug 2003, McAughan, Matt wrote: Does anyone keep a known telemarketer caller id database? If not has anyone proposed an Asterisk community project to share this information? Sort of a nation wide blacklist so Asterisk'ers can cut down on the garbage calls... ___ 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
RE: [Asterisk-Users] Asterisk crashes when trying to load G.729 module.
This is generally indicates a problem with the licensing process (which is severely flawed and full of bugs) on your server... Did you make it through the registration process OK? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Sunday, July 20, 2003 12:18 AM To: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk crashes when trying to load G.729 module. Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ 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
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
As I understand it, the playback applications need to have native access to a piece of digium hardware to perform well... Under the virtual machine, that won't happen. Matt Hardeman PaperSoft - Original Message - From: David Boreham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 2:01 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. Your e-mail made me chuckle. When I worked at Octel/Lucent in the mid-90's we were constantly sniped at for trying to make a voicemail system which ran on general purpose computers, operating systems, and message stores. It was hard work to get it to run smoothly back then even though we were the only application on the box. And today there's a guy who's trying to do the same thing in a VIRTUAL MACHINE !?!?!? sigh ___ 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
RE: [Asterisk-Users] OT: list format vs newsgroup format
I agree! phpbb is great! Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylor Sent: Friday, July 18, 2003 9:36 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT: list format vs newsgroup format IF there was a consideration for a change, I prefer: phpbb it's open source and easy to use. www.phpbb.com you can still get emails from the posts. -- Original Message -- From: Chris Earle (CBL) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 18 Jul 2003 10:02:22 -0400 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... only problem being that this list requires list membership before postingShrug C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ 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
RE: [Asterisk-Users] OT: list format vs newsgroup format
Hey there! Actually, yes, I do take 80 mg. Adderall XR daily, but I hardly think that's relevant to this thread or for that matter this mailing list. Windows on the desktop is *still* a reality... There are some things I just have to have it for... As such, here at my desk at work, I have this laptop running Windows... Shame on me. I do have quite a few linux systems, though. Like many others here, I understand the thread-view features of outlook and other mail readers... I consider them all a nasty kludge. Fancy formatting and graphics are optional. Phpbb is configurable at the user level with a number of skins, including some very simple nearly plain-text types that work quite well with text based web browsers. As for the bandwidth issue, servers migrating from a mailing list format to a forum format generally experience reductions in used bandwidth as not every user will constantly receive all messages. Users visit the site on a timeframe convenient to them, and read only the posts that interest them, by thread. As they are able to quickly reject or ignore threads that are irrelevant to them, they spend their time and bandwidth reading messages that matter to them. I didn't mention earlier that most forum software these days has a subscriber to forum feature that allows users to receive emails of the messages from the forums automatically, if they prefer. Frankly, I don't think any of the assertions in my prior email were terribly absurd, but each is entitled to his own opinion. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, July 18, 2003 12:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] OT: list format vs newsgroup format On Fri, 2003-07-18 at 12:11, Matthew Hardeman wrote: I feel your pain! I tend to prefer html based forums... They keep threads well organized, cause less overhead on the server, conserve bandwidth, etc... I understand you are a crack addict by your use of Outlook, and your opinion that fancy formating and graphics could be less bandwidth intensive than plain text. The only way you use less bandwidth is if you have fewer users reading. A mailing list broadcast the message to all users, then you can read it over and over again without incurring more transport bandwidth with the original server. They also give visual priority to new threads and active threads... What do you think bold on unread is? Get a threaded mail reader, and enjoy. On the other hand, to deal with forums, you actually have to go check them... Yep, reason not to use them on top of the above absurdities you try and put forth. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Earle (CBL) Sent: Friday, July 18, 2003 9:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] OT: list format vs newsgroup format 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... only problem being that this list requires list membership before postingShrug C h r i sE a r l e System Solutions Specialist ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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
RE: [Asterisk-Users] Cisco 7960
Ive run into this before, and its a pain to debug Be sure that your eth0 interface (primary, first interface) is set to your internal address space (of the same subnet that you assign to the phone). You can add an IP alias on eth0:1 if you need an external IP on that box as well, but you must have them in that order: internal = eth0, externals, others eth0:1+ Try that Matt PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Carlson Sent: Thursday, July 17, 2003 6:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 I bought a 7960 it was running version 3.3 of the SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the ethernet it doesn't rebootor if I remove all the lines in the SIP config it won't reboot. Since this is used cisco won't give me any support. For now I am running the MGCP version but eh asterisk seems to have some issues with it. Thanks, Will
RE: [Asterisk-Users] Any dialing tricks...
Hey Kim! I used to run that scam myself! You go! Back in the day, actually, I had our legacy lucent merlin phone system wired up to a modem on a webserver which could config it And with some voicemail tricks and the like, it was possible for me to visit a little WAP site on my phone, and have it dial the number and bridge the call to me In asterisk it would be a lot more graceful You could build a little script to look for a two-way message from you and use the outbound call spool to set up a call Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis Sent: Thursday, July 17, 2003 10:38 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Any dialing tricks... Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or maybe have a call back with a DISA and then just dial my phone number I am trying to reach Just a thought! Kim Callis
RE: [Asterisk-Users] Cisco 7960
Heres a hint Upgrade back to 5.1. It still has bugs, but theyre no worse than 5.0 Put it in the same subnet as the asterisk server, and make sure that the eth0 interface (the FIRST interface in the box) is on that subnet. Generally, you will want this to be your internal address space. 192.168.0.x If you asterisk server needs an external IP on the same Ethernet, just do an alias as eth0:1 or something along those lines. Make the primary hostname of the server (as reflected by the hostname command) match up in /etc/hosts as the internal IP address. Asterisk apparently picks up the first IP address on the system to use as its source IP address for all things SIP If you have a Cisco phone communicate with Asterisk on another subnet known to the system (via an IP alias on the same Ethernet card), Ive found that the Cisco 7960 will crash and burn. I suspect that if Asterisk were modified to source the communications back over the interface it received them, the crash would no longer happen. Check your configuration files Ive had these phones crash on me before if your networking isnt very friendly to them, but never before just during the booting sequence Trust me; you can get this phone to work Its just a matter of patience and experimenting, and lots of free time wasted on Ethereal J As an aside, Ive actually been actively working with a Cisco developer (even today) to generate more debug information for them on the network caused crash and reboot issue, and they think theyve about got it licked I believe they will be sending me a firmware image to test soon that will have at least that bug, and probably more, fixed. Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Carlson Sent: Thursday, July 17, 2003 5:10 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 Cisco's website has some stuff on there website which seems to indicate if the 7960 cannot contact the call manager server it reboots. However to my knowledge this has never had call manager software before and cisco doesn't mention this feature with the SIP firmware. I downgraded to 5.0 unfortunately due to only being able to run Secure images now thats as far back as I can go. Thanks again cisco for this feature. From what I can tell the phone never talked to the Asterisk box. If I turn on SIP debugging I do not see any traffic coming from the cisco box. Although I did have them on seperate subnets. Let me try putting them on the same subnet and see if that helps. Thanks, Will - Original Message - From: Matthew Hardeman To: [EMAIL PROTECTED] Sent: Thursday, July 17, 2003 12:58 PM Subject: RE: [Asterisk-Users] Cisco 7960 Ive run into this before, and its a pain to debug Be sure that your eth0 interface (primary, first interface) is set to your internal address space (of the same subnet that you assign to the phone). You can add an IP alias on eth0:1 if you need an external IP on that box as well, but you must have them in that order: internal = eth0, externals, others eth0:1+ Try that Matt PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Carlson Sent: Thursday, July 17, 2003 6:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 I bought a 7960 it was running version 3.3 of the SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the ethernet it doesn't rebootor if I remove all the lines in the SIP config it won't reboot. Since this is used cisco won't give me any support. For now I am running the MGCP version but eh asterisk seems to have some issues with it. Thanks, Will
RE: [Asterisk-Users] G729 quality
In my experience it depends on who or what you are speaking G729 to, but yes, generally... Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Rychter Sent: Tuesday, July 15, 2003 7:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 quality This is a MIME-formatted message. If you see this text it means that your E-mail software does not support MIME-formatted messages. --=_megabox.papersoft.com-18016-1058316735-0001-2 Content-Transfer-Encoding: quoted-printable Does G.729 provide better voice quality than GSM? (a question for people who have tried both) =2D-J. --=_megabox.papersoft.com-18016-1058316735-0001-2 Content-Type: application/pgp-signature Content-Transfer-Encoding: 7bit -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQA/FJY0Lth4/7/QhDoRAmk0AJ45/gMclZZ+Ddvcvff0LrHk1vlV9wCffy7U JXzzyasgBE+d2xsxniFfKeA= =LKkp -END PGP SIGNATURE- --=_megabox.papersoft.com-18016-1058316735-0001-2-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EZ-Install
Maybe it's just me... But I fail to see the reasoning behind branching to a whole new distribution just to support an easy, out of the box Asterisk install. Perhaps just the creation of an RPM package with a basic configuration would be the ticket? The one potential exception to this would be if you wrote a distribution with advanced hardware detection and preconfiguration such that during the install process, Digium hardware is detected and you can go ahead and configure spans and channels, etc. In that case, the distribution might have some unique value. Short of that, I cannot imagine a new distribution just to package together a pre-configured Asterisk configuration. Even if you wrote an installation process like that, couldn't it be just as well implemented with a clever RPM-based installation and some nice plain old userspace configuration tools? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylor Sent: Monday, July 14, 2003 11:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] EZ-Install Not CD based. Just CD install. When you reboot Linux with asterisk is installed. You could add any other tools you think are necessary. User then just does config. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 14 Jul 2003 10:18:24 -0500 On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ 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] Hardware Vendors
Hi All! Can anyone direct me to any websites / manufacturers out there who are making small, put-it-in-the-closet-and-forget-it type systems for building routers, home gateway servers, that sort of thing? My fantasy machine for this purpose would be along the lines of a mini-itx system with external power supply, dual Ethernet interfaces on board, and one PCI slot available. If it had one real serial port on it, that would be great too. Am I dreaming, or does it exist for a reasonable price? I would be willing to go the 500 MHz 1 GHz range. Something without a fan would be really nice. Im basically looking for a system that someone out there is stamping out in quantities and isnt too outrageous in price. Does it exist, and if so who sells it? It seems to me a system like the above described would be perfect for building out a home gateway / home asterisk server Matt Hardeman PaperSoft
RE: [Asterisk-Users] G729 licensing
Missing something? No... So far as I'm aware there is no freely available G729 codec available that will run under Linux... Kind of funny that there *is* one for Windows, isn't it? As an aside, though, what kind of equipment are you using, and what circumstances are you communicating in? ALAW ULAW make great codecs for use on a LAN. :) I've also had great luck using GSM with the Snom200 running the very latest firmware. (1.18s) It's not yet posted on their website, but they will give you a link to it should you write their support team... Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Rychter Sent: Monday, July 14, 2003 12:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 licensing This is a MIME-formatted message. If you see this text it means that your E-mail software does not support MIME-formatted messages. --=_megabox.papersoft.com-1525-1058205320-0001-2 Content-Transfer-Encoding: quoted-printable 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? =2D-J. --=_megabox.papersoft.com-1525-1058205320-0001-2 Content-Type: application/pgp-signature Content-Transfer-Encoding: 7bit -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQA/EujILth4/7/QhDoRAty1AJ9z5xm6Zkj/oiYkm1buSfjceuC2UQCgh91y FqD3wdGNWgDXjrZLPD5nkcY= =SUZa -END PGP SIGNATURE- --=_megabox.papersoft.com-1525-1058205320-0001-2-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones
chan_sccp would be nice :) I've been playing around with the 7960's and have really enjoyed the 7960 as a desktop phone. It's physically well constructed, has a sturdy/heavy handset (a good thing in my book), a very pleasant user interface... And if you're willing to make changes to your network setup to accomodate it's presently finicky firmwarre, you'll be ok... The firmware issue for the 7960 SIP is yet to be resolved, but hopefully it'll come around... I think the entire 79XX lines of phones by Cisco has lots of promise, but we won't really see the others (than the 7960/7940) be much use in Asterisk until there is native support in Asterisk for sccp... I did hear a rumor that someone was working on it... Matt Hardeman PaperSoft - Original Message - From: John Todd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 3:32 PM Subject: [Asterisk-Users] Fwd:[Vocal] Question about Cisco IP hard phones Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the very inexpensive Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to appear. ;-) JT Reply-To: [EMAIL PROTECTED] From: David Kelly [EMAIL PROTECTED] To: Chok Lam [EMAIL PROTECTED], [EMAIL PROTECTED] Org [EMAIL PROTECTED] Subject: RE: [Vocal] Question about Cisco IP hard phones Date: Mon, 14 Jul 2003 11:56:45 -0700 Folks, For the time being, the low-end Cisco IP phones, 7902G and 7912G support SCCP only. The 7905G supports both H.323 and SCCP, but we are not prioritizing new development on the H.323 load. This load is a legacy from the 7905 phone that was released in 2003 and EOL'd last week. This autumn, we will release a SIP image for the 7905G and 7912G. There are no plans to release a SIP image for the 7902G. David [snip] ___ 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
Re: [Asterisk-Users] G729 licensing
I'm not familiar with the codec support in Gnomeeting, but have you tried a codec like iLBC? I had great success running ilbc over IAX2 between my home and office. Matt Hardeman PaperSoft - Original Message - From: Jan Rychter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 14, 2003 6:44 PM Subject: Re: [Asterisk-Users] G729 licensing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960s
Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I discovered something interesting... By screwing around with both the source IP address of a SIP message, along with certain IP addresses in the SIP message itself, it's quite easy to crash the Cisco. In short, it would betrivial to DOS (by forcing continuous crashes and the subsequent reboots) any Cisco 7960 that you can route UDP packets to... Matt HardemanPaperSoft
Re: [Asterisk-Users] Cisco 7960s
I have an open ticket at cisco with status development review; workaround provided. I'm going to remind them of the potential security consequences later today... The tech I've been working with seems very competent, and I suspect this may eventually get dealt with... Matt Hardeman PaperSoft - Original Message - From: Josh Howlett [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 11, 2003 3:30 AM Subject: Re: [Asterisk-Users] Cisco 7960s Cisco and bugtraq need to know this! josh. On Fri, 2003-07-11 at 09:21, Matthew Hardeman wrote: Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I discovered something interesting... By screwing around with both the source IP address of a SIP message, along with certain IP addresses in the SIP message itself, it's quite easy to crash the Cisco. In short, it would be trivial to DOS (by forcing continuous crashes and the subsequent reboots) any Cisco 7960 that you can route UDP packets to... Matt Hardeman PaperSoft -- --- Josh Howlett, Networking Digital Communications, Information Systems Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: [EMAIL PROTECTED] ___ 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
Re: [Asterisk-Users] Weird experience with MOH
If you're on a RedHat system, mpg321 is installed by default, and is symlinked to as mpg123... So, it can easily look like you have mpg123, but you really have mpg321... Sorry if you checked for that, and I've offended, but just thought I'd offer. Matt Hardeman PaperSoft - Original Message - From: BK [address only for mailing lists] [EMAIL PROTECTED] To: Asterisk List [EMAIL PROTECTED] Sent: Friday, July 11, 2003 10:09 PM Subject: [Asterisk-Users] Weird experience with MOH Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there was only silence when putting a caller on hold. Somebody told me I may have the wrong mp3 app (321 vs 123) while I was getting busy with something else and so I put this aside. Although I found that I did have mpg123 installed. Yesterday, I copied some music files into /var/lib/asterisk/mohmp3 in anticipation that I would get this to work eventually and to my surprise, putting a caller on hold now plays the music. I have no idea why it didn't work at first, but it would seem that for some unknown reason, Asterisk didn't like the sole sample-hold.mp3 file. rgds bk ___ 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] Cisco 7960 SIP Craziness...
Hi All! First, let me introduce myself, as this is my first post to the list (Ive been lurking for quite some time now). My name is Matt Hardeman, and I work for a software development firm in Birmingham, AL. We are interested in the Asterisk PBX and its various configurations first as an internal solution for our occasionally bizarre telephony needs, and eventually are interested in potentially working with Asterisk commercially and building services around it Anyway, at this state in the game I am currently playing with my Asterisk configuration with various SIP devices, trying to find the ultimate desktop UI Currently, Ive purchased a Cisco 7960 which unfortunately comes pre-installed with a CallManager image. No problem, or so I thought The reseller sent me the latest SIP firmware I have set up my TFTP server, and the phone talks to it, downloads is OS79XX.TXT file, etc, etc. When it goes to download the firmware image, it fails the download, and keeps repeatedly trying. This scenario is covered in the Cisco FAQ for converting a CallManager 7960 to SIP It essentially is a bug in the firmware on the phone which requires upgrading to an intermediary, older SIP firmware first. URL: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml I dont have this, and my reseller doesnt have it handy either, though they promise to get it (some day?) At any rate, Ive opened a Cisco tech support incident with hopes that theyll be able to provide me the files quick and easy. But, that failing, can anyone out there send me P0S30200.bin??? Thanks, Matt Hardeman PaperSoft
